plugin/adin_oss.c

説明を見る。

/***************************************************************************/

#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include "plugin_defs.h"

#define PLUGIN_TITLE "A/D-in plugin for Julius"

#define INPUT_OPT "myadin"

int
initialize()
{
  return 0;
}

int
get_plugin_info(int opcode, char *buf, int buflen)
{
  switch(opcode) {
  case 0:
    /* plugin description string */
    strncpy(buf, PLUGIN_TITLE, buflen);
    break;
  }

  return 0;
}

/************************************************************************/
/************************************************************************/
/* A/D-in plugin functions */

void
adin_get_optname(char *buf, int buflen)
{
  strncpy(buf, INPUT_OPT, buflen);
}

int
adin_get_configuration(int opcode)
{
  /* For your convenience, UNCOMMENT ONE OF THEM BELOW that match your needs */

  /* typical values for live microphone/line input */
  switch(opcode) {
  case 0:       
    return 1;
  case 1:
    return 1;
  case 2:
    return 1;
  }
  /* typical values for offline file input */
  /* 
   * switch(opcode) {
   * case 0:       
   *   return 0;
   * case 1:
   *   return 0;
   * case 2:
   *   return 0;
   * }
   */
  /* typical setting for tcpip input */
  /* assuming speech to be segmented at sender */
  /* 
   * switch(opcode) {
   * case 0:       
   *   return 1;
   * case 1:
   *   return 0;
   * case 2:
   *   return 0;
   * }
   */
  /* typical setting for tcpip input */
  /* assuming receiving continous speech stream and segmented
     should be done at Julius side */
  /* 
   * switch(opcode) {
   * case 0:       
   *   return 1;
   * case 1:
   *   return 1;
   * case 2:
   *   return 0;
   * }
   */
}
 

/************************************************************************/

#include <sys/types.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <sys/soundcard.h>
static int audio_fd;
static int freq;

boolean
adin_standby(int sfreq, void *dummy)
{
  /* store the frequency */
  freq = sfreq;
  return TRUE;
}
 
boolean
adin_open()
{
  /* do open the device */
  int fmt;
  int stereo;
  int ret;
  int s;
  char buf[2];

  if ((audio_fd = open("/dev/dsp", O_RDONLY)) == -1) {
    printf("Error: cannot open /dev/dsp\n");
    return FALSE;
  }
  fmt = AFMT_S16_LE;               /* 16bit signed (little endian) */
  if (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &fmt) == -1) {
    printf("Error: failed set format to 16bit signed\n");
    return FALSE;
  }
  stereo = 0;                   /* mono */
  ret = ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo);
  if (ret == -1 || stereo != 0) {
    stereo = 1;
    ret = ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &stereo);
    if (ret == -1 || stereo != 1) {
      printf("Error: failed to set monoral channel\n");
      return FALSE;
    }
  }
  s = freq;
  if (ioctl(audio_fd, SNDCTL_DSP_SPEED, &s) == -1) {
    printf("Erorr: failed to set sample rate to %dHz\n", freq);
    return FALSE;
  }

  /* start recording... */
  read(audio_fd, buf, 2);

  return(TRUE);
}

int
adin_read(SP16 *buf, int sampnum)
{
  audio_buf_info info;
  int size, cnt;

  /* get sample num that can be read without blocking */
  if (ioctl(audio_fd, SNDCTL_DSP_GETISPACE, &info) == -1) {
    printf("Error: adin_oss: failed to get number of samples in the buffer\n");
    return(ADIN_ERROR);
  }
  /* get them as much as possible */
  size = sampnum * sizeof(SP16);
  if (size > info.bytes) size = info.bytes;
  size &= ~ 1;          /* Force 16bit alignment */
  cnt = read(audio_fd, buf, size);
  if ( cnt < 0 ) {
    printf("Error: adin_oss: failed to read samples\n");
    return (ADIN_ERROR);
  }
  cnt /= sizeof(short);
    
  return(cnt);
}

boolean
adin_close()
{
  close(audio_fd);
  return TRUE;
}

/************************************************************************/

boolean
adin_terminate()
{
  printf("terminate request\n");
  return TRUE;
}

boolean
adin_pause()
{
  printf("pause request\n");
  return TRUE;
}

boolean
adin_resume()
{
  printf("resume request\n");
  return TRUE;
}

/* end of file */

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