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 <alsa/asoundlib.h>
b,编程录音代码.
*/ /* Use the newer ALSA API */ #define ALSA_PCM_NEW_HW_PARAMS_API #include <stdio.h> #include <stdlib.h> #include <alsa/asoundlib.h> #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 ", 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 ", 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 "); snd_pcm_prepare(handle); } else if (rc < 0) { fprintf(stderr, "error from read: %s ", snd_strerror(rc)); } else if (rc != (int)frames) { fprintf(stderr, "short read, read %d frames ", rc); } if(write(fd_f, buffer, size)==-1) { perror("write to sound wrong!!"); } if (rc != size) fprintf(stderr, "short write: wrote %d bytes ", 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 <unistd.h> #include <fcntl.h> #include <sys/types.h> #include <sys/stat.h> #include <sys/ioctl.h> #include <stdlib.h> #include <stdio.h> #include <string.h> #include <linux/soundcard.h> #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; }