[開發記錄] 函式庫調用 - 使用ALSA進行音訊資料擷取 之其一

目前打算使用ALSA進行資料擷取，

關於ALSA API所搜尋到的網路資料( http://equalarea.com/paul/alsa-audio.html )，

其中有一段 Mini Capture程式，並作了些許修改，如底下所示：

//---------------ALSA 測試程式---------------#include <stdio.h>#include <stdlib.h>#include <alsa/asoundlib.h>      int main (int argc, char *argv[]){int i;int err;unsigned int sample_rate;int sample_dir;int size;snd_pcm_uframes_t frames;char *buf;snd_pcm_t *capture_handle;snd_pcm_hw_params_t *hw_params;if ((err = snd_pcm_open(&capture_handle, argv[1], SND_PCM_STREAM_CAPTURE, 0)) < 0) {fprintf (stderr, "cannot open audio device %s (%s)\n",  argv[1], snd_strerror (err));exit(1);}printf("open OK!\n");   if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0) {fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n", snd_strerror (err));exit(1);}printf("set allocation OK!\n"); if ((err = snd_pcm_hw_params_any(capture_handle, hw_params)) < 0) {fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n", snd_strerror (err));exit(1);}printf("set para OK!\n");if ((err = snd_pcm_hw_params_set_access(capture_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {fprintf (stderr, "cannot set access type (%s)\n", snd_strerror (err));exit(1);}printf("set access OK!\n");if ((err = snd_pcm_hw_params_set_format(capture_handle, hw_params, SND_PCM_FORMAT_S16_LE)) < 0) {fprintf (stderr, "cannot set sample format (%s)\n", snd_strerror (err));exit(1);}printf("set sample format OK!\n");sample_rate=44100;sample_dir=0;//snd_pcm_hw_params_set_rate_near (snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int *val, int *dir) if ((err = snd_pcm_hw_params_set_rate_near(capture_handle, hw_params, &sample_rate, &sample_dir)) < 0) {fprintf (stderr, "cannot set sample rate (%s)\n", snd_strerror (err));exit(1);}printf("set sample rate OK!\n");if ((err = snd_pcm_hw_params_set_channels(capture_handle, hw_params, 2)) < 0) {fprintf (stderr, "cannot set channel count (%s)\n", snd_strerror (err));exit(1);}printf("set channel OK!\n");if ((err = snd_pcm_hw_params(capture_handle, hw_params)) < 0) {fprintf (stderr, "cannot set parameters (%s)\n", snd_strerror (err));exit(1);}printf("set para OK!\n");snd_pcm_hw_params_free(hw_params);if ((err = snd_pcm_prepare (capture_handle)) < 0) {fprintf (stderr, "cannot prepare audio interface for use (%s)\n", snd_strerror (err));exit(1);}printf("prepare audio interface OK!\n");frames=128;size=frames*4;buf=(char*)malloc(size);for (i = 0; i < 10; ++i) {//snd_pcm_readi(snd_pcm_t *pcm,void *buffer,snd_pcm_uframes_t size)if ((err = snd_pcm_readi (capture_handle, buf, 128)) != 128) {fprintf (stderr, "read from audio interface failed (%s)\n", snd_strerror (err));exit (1);}}free(buf);snd_pcm_close (capture_handle);printf("end of Mini-Capture!\n");exit (0);}

在編輯時遭遇零件庫問題，直接將 linux-devkit/arm-none-linux-guneabi/usr/include以及lib都複製到 CodeSourcery/Sourcery_G++_Lite/arm-none-linux-guneabi/相對應的資料夾，解決!

使用 gcc時，需要指定使用 -lasound 函式庫。

程式內，部分增加註解的地方，是由於函式庫的參數錯誤，出現Segmentation fail的訊息，修改參數類型後重新編譯，

在BBxM內，執行./MiniCapture default 指定設備使用default值，就可進行截取聲音訊號的工作，

內部參數的設定意義，等我了解後在另外補上。