libsent/src/adin/adin_mic_freebsd.c

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/*
 * Copyright (c) 1991-2007 Kawahara Lab., Kyoto University
 * Copyright (c) 2000-2005 Shikano Lab., Nara Institute of Science and Technology
 * Copyright (c) 2005-2007 Julius project team, Nagoya Institute of Technology
 * All rights reserved
 */

/* Thanks to Kentaro Nagatomo for information */
/* All functions are the same as OSS version, except the header filename */

#include <sent/stddefs.h>
#include <sent/adin.h>

#include <sys/ioctl.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <poll.h>

/* sound header */
#ifdef HAVE_SYS_SOUNDCARD_H
#include <sys/soundcard.h>
#elif HAVE_MACHINE_SOUNDCARD_H
#include <machine/soundcard.h>
#endif

#define DEFAULT_DEVICE "/dev/dsp"

static int audio_fd;            
static boolean need_swap;       
struct pollfd fds[1];           

#define FREQALLOWRANGE 200      
#define POLLINTERVAL 200        


boolean
adin_mic_standby(int sfreq, void *arg)
{
  int fmt, fmt_can, fmt1, fmt2, rfmt; /* sampling format */
  int samplerate;               /* actual sampling rate */
  int stereo;           /* mono */
  char *defaultdev = DEFAULT_DEVICE; /* default device */
  char *devname;

  /* set device name */
  if ((devname = getenv("AUDIODEV")) == NULL) {
    devname = defaultdev;
    jlog("Stat: adin_freebsd: device name = %s\n", devname);
  } else {
    jlog("Stat: adin_freebsd: device name obtained from AUDIODEV: %s\n", devname);
  }
  
  /* open device */
  if ((audio_fd = open(devname, O_RDONLY)) == -1) {
    jlog("Error: adin_freebsd: failed to open %s\n", devname);
    return(FALSE);
  }

  /* check whether soundcard can record 16bit data */
  /* and set fmt */
  if (ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &fmt_can) == -1) {
    jlog("Error: adin_freebsd: failed to get formats from audio device\n");
    return(FALSE);
  }
#ifdef WORDS_BIGENDIAN
  fmt1 = AFMT_S16_BE;
  fmt2 = AFMT_S16_LE;
#else
  fmt1 = AFMT_S16_LE;               /* 16bit signed (little endian) */
  fmt2 = AFMT_S16_BE;               /* (big endian) */
#endif /* WORDS_BIGENDIAN */
  /* fmt2 needs byte swap */
  if (fmt_can & fmt1) {
    fmt = fmt1;
    need_swap = FALSE;
  } else if (fmt_can & fmt2) {
    fmt = fmt2;
    need_swap = TRUE;
  } else {
    jlog("Error: adin_freebsd: 16bit recording not supported on this device\n");
    return FALSE;
  }
#ifdef DEBUG
  if (need_swap) {
    jlog("Stat: adin_freebsd: samples need swap\n");
  } else {
    jlog("Stat: adin_freebsd: samples need not swap\n");
  }
#endif
  
  if (close(audio_fd) != 0) return FALSE;

  /* re-open for recording */
  /* open device */
  if ((audio_fd = open(devname, O_RDONLY)) == -1) {
    jlog("Error: adin_freebsd: failed to open %s", devname);
    return(FALSE);
  }
  /* set format, samplerate, channels */
  rfmt = fmt;
  if (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &rfmt) == -1) {
    jlog("Error: adin_freebsd: failed to get available formats from device\n");
    return(FALSE);
  }
  if (rfmt != fmt) {
    jlog("Error: adin_freebsd: 16bit recording is not supported on this device\n");
    return FALSE;
  }

  stereo = 0;                   /* mono */
  if (ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo) == -1) {
    jlog("Error: adin_freebsd: failed to set monoral recording\n");
    return(FALSE);
  }
  if (stereo != 0) {
    jlog("Error: adin_freebsd: monoral recording not supported on this device\n");
    return FALSE;
  }

  samplerate = sfreq;
  if (ioctl(audio_fd, SNDCTL_DSP_SPEED, &samplerate) == -1) {
    jlog("Erorr: adin_freebsd: failed to set sample rate to %dHz\n", sfreq);
    return(FALSE);
  }
  if (samplerate < sfreq - FREQALLOWRANGE || samplerate > sfreq + FREQALLOWRANGE) {
    jlog("Error: adin_freebsd: failed to set sampling rate to near %dHz. (%d)\n", sfreq, samplerate);
    return FALSE;
  }
  if (samplerate != sfreq) {
    jlog("Warning: adin_freebsd: specified sampling rate was %dHz but set to %dHz, \n", sfreq, samplerate);
  }

  /* set polling status */
  fds[0].fd = audio_fd;
  fds[0].events = POLLIN;
  
  return TRUE;
}
 
boolean
adin_mic_begin()
{
  return(TRUE);
}

boolean
adin_mic_end()
{
  /*
   * Not reset device on each end of speech, just let the buffer overrun...
   * Resetting and restarting of recording device sometimes causes
   * hawling noises at the next recording.
   * I don't now why, so take the easy way... :-(
   */
  return TRUE;
}

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

  /* wait till at least one sample can be read */
  poll(fds, 1, POLLINTERVAL);
  /* get actual sample num in the device buffer */
  if (ioctl(audio_fd, SNDCTL_DSP_GETISPACE, &info) == -1) {
    jlog("Error: adin_freebsd: adin_mic_read: sndctl_dsp_getispace");
    return(-2);
  }
  
  /* get them as much as possible */
  size = sampnum * sizeof(SP16);
  if (size > info.bytes) size = info.bytes;
  cnt = read(audio_fd, buf, size);
  if ( cnt < 0 ) {
    jlog("Error: adin_freebsd: adin_mic_read: read error\n");
    return ( -2 );
  }
  cnt /= sizeof(short);
  if (need_swap) swap_sample_bytes(buf, cnt);
  return(cnt);
}

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