Linux聲卡錄音程序之——mp3(通過ALS…

此程序通過ALSA打開聲卡設備，和從聲卡讀取數據，避免直接操作聲卡。由於不同主機，不同聲卡的名字可能不一樣，直接通過設備名操作聲卡有侷限性，故改進成通過ALSA操作。



#include "lame.h"
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <fcntl.h>
#include <errno.h>
//#include <signal.h>
//#include <sys/ioctl.h>
#include <memory.h>
//#include <linux/soundcard.h>
#include <alsa/asoundlib.h>
#include <sys/time.h>

#define TIMES    10   //錄音時間,秒
#define RATE    41000 //採樣頻率
#define BITS    16   //量化位數
#define CHANNELS 2   //聲道數目
#define INBUFF_SIZE 4096
#define MP3BUFF_SIZE (int) (1.25 * INBUFF_SIZE) + 7200

// handle the case of underrun or overrun
int xrun(snd_pcm_t* handle);
// handle the case that device busy
int suspend(snd_pcm_t* handle);
// set alsa params
int SetFormat(snd_pcm_t* handle, unsigned int channels, unsigned int rate)
{
    int ret = 0;
   
    ret = snd_pcm_set_params(handle, SND_PCM_FORMAT_S16_LE, SND_PCM_ACCESS_RW_INTERLEAVED, channels, rate, 1, 500000);
    if (ret != 0) {
        printf("Unable to set params,error(%s)\n", snd_strerror(ret));
    }

    return ret;
}

int main(int argc, char **argv)
{
    int fd_dsp;
    FILE* fd_tmp;
    FILE* fd_mp3;
    lame_global_flags* gfp;
    short *input_buff;
    unsigned char *mp3_buff;

    int samples;
    int mp3_bytes;
    int write_bytes;
    int num = 0;
    int ch;
    int i = 0;
    int ret = 0;
    snd_pcm_t* handle;

    if (argc != 2) {
        fprintf(stderr, "Useage: ./mp3_record test.mp3\n");
        return -1;
    }

    if (snd_pcm_open(&handle, "default", SND_PCM_STREAM_CAPTURE, 0) != 0) {
        printf("Failed to open device\n");
        return -1;
    }
    if (SetFormat(handle, CHANNELS, RATE) != 0) {
        printf("Cannot set sound device in bit 16, channel 2, speed 44100.\n");
        return -1;
    }

    if ((fd_mp3 = fopen(argv[1], "w")) == NULL) {
        fprintf(stderr, "Open file error: %s\n", strerror(errno));
        ret = -1;
        goto CLOSE_DSP;
    }
    gfp = lame_init();
    if (gfp == NULL) {
        printf("lame_init failed\n");
        ret = -1;
        goto CLOSE_MP3;
    }

    lame_set_in_samplerate(gfp, RATE);
    //lame_set_out_samplerate(gfp, RATE);
    lame_set_num_channels(gfp, CHANNELS);
    //lame_set_brate(gfp, 24);
    //lame_set_VBR_min_bitrate_kbps(gfp, lame_get_brate(gfp));
    //lame_set_quality(gfp,7);
    ret = lame_init_params(gfp);
    if (ret < 0) {
        printf("lame_init_params returned %d\n", ret);
        ret = -1;
        goto CLOSE_LAME;
    }
    ch = lame_get_num_channels(gfp);
    printf("default rate is %d, out rate is %d.\n", lame_get_in_samplerate(gfp), lame_get_out_samplerate(gfp));