libsent/src/adin/adin_mic_linux_oss.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
 */

#include <sent/stddefs.h>

#include <sent/adin.h>
#include <sys/ioctl.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <sys/time.h>
#include <fcntl.h>

/* sound header */
#ifdef HAS_OSS
#if defined(HAVE_SYS_SOUNDCARD_H)
#include <sys/soundcard.h>
#elif defined(HAVE_MACHINE_SOUNDCARD_H)
#include <machine/soundcard.h>
#endif
#endif

#define DEFAULT_DEVICE "/dev/dsp"

#define FREQALLOWRANGE 200      
#define MAXPOLLINTERVAL 300     

#define MAX_FRAGMENT_MSEC 50

#define MIN_FRAGMENT_SIZE 256

static int audio_fd;            
static boolean need_swap;       
static int frag_size;           
static boolean stereo_rec;      

boolean
adin_oss_standby(int sfreq, void *dummy)
{
#ifndef HAS_OSS
  jlog("Error: OSS not compiled in\n");
  return FALSE;
#else
  int fmt, fmt_can, fmt1, fmt2, rfmt; /* sampling format */
  int samplerate;       /* 16kHz */
  char *defaultdev = DEFAULT_DEVICE; /* default device */
  char *devname;
  int frag;
  int frag_msec;
  char *env;

  /* set device name */
  if ((devname = getenv("AUDIODEV")) == NULL) {
    devname = defaultdev;
    jlog("Stat: adin_oss: device name = %s\n", devname);
  } else {
    jlog("Stat: adin_oss: device name obtained from AUDIODEV: %s\n", devname);
  }

  /* open device */
  if ((audio_fd = open(devname, O_RDONLY|O_NONBLOCK)) == -1) {
    jlog("Error: adin_oss: 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_oss: 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_oss: 16bit recording not supported on this device\n");
    return FALSE;
  }
#ifdef DEBUG
  if (need_swap) {
    jlog("Stat: adin_oss: samples need swap\n");
  } else {
    jlog("Stat: adin_oss: 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_oss: failed to open %s", devname);
    return(FALSE);
  }

  /* try to set a small fragment size to minimize delay, */
  /* although many devices use static fragment size... */
  /* (and smaller fragment causes busy buffering) */
  {
    int arg;
    int f, f2;

    /* if environment variable "LATENCY_MSEC" is defined, try to set it
       as a minimum latency in msec (will be rouneded to 2^x). */
    if ((env = getenv("LATENCY_MSEC")) == NULL) {
      frag_msec = MAX_FRAGMENT_MSEC;
    } else {
      frag_msec = atoi(env);
    }
      
    /* get fragment size from MAX_FRAGMENT_MSEC and MIN_FRAGMENT_SIZE */
    f = 0;
    f2 = 1;
    while (f2 * 1000 / (sfreq * sizeof(SP16)) <= frag_msec
           || f2 < MIN_FRAGMENT_SIZE) {
      f++;
      f2 *= 2;
    }
    frag = f - 1;

    /* set to device */
    arg = 0x7fff0000 | frag;
    if (ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &arg)) {
      jlog("Stat: adin_oss: set fragment size to 2^%d=%d bytes (%d msec)\n", frag, 2 << (frag-1), (2 << (frag-1)) * 1000 / (sfreq * sizeof(SP16)));
    }
  }
  
  /* set format, samplerate, channels */
  rfmt = fmt;
  if (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &rfmt) == -1) {
    jlog("Error: adin_oss: failed to get available formats from device\n");
    return(FALSE);
  }
  if (rfmt != fmt) {
    jlog("Error: adin_oss: 16bit recording is not supported on this device\n");
    return FALSE;
  }

  {
    /* try SNDCTL_DSP_STEREO, SNDCTL_DSP_CHANNELS, monaural, stereo */
    int channels;
    int stereo;
    boolean ok_p = FALSE;

    stereo = 0;                 /* mono */
    if (ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo) == -1) {
      /* failed: SNDCTL_DSP_STEREO not supported */
      jlog("Stat: adin_oss: sndctl_dsp_stereo not supported, going to try another...\n");
    } else {
      if (stereo != 0) {
        /* failed to set monaural recording by SNDCTL_DSP_STEREO */
        jlog("Stat: adin_oss: failed to set monaural recording by sndctl_dsp_stereo\n");
        jlog("Stat: adin_oss: going to try another...\n");
      } else {
        /* succeeded to set monaural recording by SNDCTL_DSP_STEREO */
        //jlog("Stat: adin_oss: recording now set to mono\n");
        stereo_rec = FALSE;
        ok_p = TRUE;
      }
    }
    if (! ok_p) {               /* not setup yet */
      /* try using sndctl_dsp_channels */
      channels = 1;
      if (ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &channels) == -1) {
        /* failed: SNDCTL_DSP_CHANNELS not supported */
        jlog("Stat: adin_oss: sndctl_dsp_channels not supported, try another...\n");
      } else {
        if (channels != 1) {
          /* failed to set monaural recording by SNDCTL_DSP_CHANNELS */
          jlog("Stat: adin_oss: failed to set monaural recording by sndctl_dsp_channels\n");
          jlog("Stat: adin_oss: going to try another...\n");
        } else {
          /* succeeded to set monaural recording by SNDCTL_DSP_CHANNELS */
          //jlog("Stat: adin_oss: recording now set to mono\n");
          stereo_rec = FALSE;
          ok_p = TRUE;
        }
      }
    }
    if (! ok_p) {
      /* try using stereo input */
      jlog("Warning: adin_oss: failed to setup monaural recording, trying to use the left channel of stereo input\n");
      stereo = 1;                       /* stereo */
      if (ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo) == -1) {
        /* failed: SNDCTL_DSP_STEREO not supported */
        jlog("Stat: adin_oss: failed to set stereo input using sndctl_dsp_stereo\n");
      } else {
        if (stereo != 1) {
          /* failed to set stereo recording by SNDCTL_DSP_STEREO */
          jlog("Stat: adin_oss: failed to set stereo input using sndctl_dsp_stereo\n");
        } else {
          /* succeeded to set stereo recording by SNDCTL_DSP_STEREO */
          jlog("Stat: adin_oss: recording now set to stereo, using left channel\n");
          stereo_rec = TRUE;
          ok_p = TRUE;
        }
      }
    }
    if (! ok_p) {               /* not setup yet */
      /* try using stereo input with sndctl_dsp_channels */
      channels = 2;
      if (ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &channels) == -1) {
        /* failed: SNDCTL_DSP_CHANNELS not supported */
        jlog("Stat: adin_oss: failed to set stereo input using sndctl_dsp_channels\n");
      } else {
        if (channels != 2) {
          /* failed to set stereo recording by SNDCTL_DSP_CHANNELS */
          jlog("Stat: adin_oss: failed to set stereo input using sndctl_dsp_channels\n");
        } else {
          /* succeeded to set stereo recording by SNDCTL_DSP_CHANNELS */
          jlog("Stat: adin_oss: recording now set to stereo, using left channel\n");
          stereo_rec = TRUE;
          ok_p = TRUE;
        }
      }
    }
    if (! ok_p) {               /* all failed */
      jlog("Error: adin_oss: failed to setup recording channels\n");
      return FALSE;
    }
  }

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

  /* get actual fragment size */
  if (ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &frag_size) == -1) {
    jlog("Error: adin_oss: failed to get fragment size\n");
    return(FALSE);
  }
  if (env == NULL) {
    jlog("Stat: adin_oss: going to set latency to %d msec\n", frag_msec);
  } else {
    jlog("Stat: adin_oss: going to set latency to %d msec (from env LATENCY_MSEC)\n", frag_msec);
  }
  jlog("Stat: adin_oss: audio I/O Latency = %d msec (fragment size = %d samples)\n", frag_size * 1000/ (sfreq * sizeof(SP16)), frag_size / sizeof(SP16));

  return TRUE;

#endif /* HAS_OSS */
}
 
boolean
adin_oss_begin()
{
  char buf[4];
  /* Read 1 sample (and ignore it) to tell the audio device start recording.
     (If you knows better way, teach me...) */
  if (stereo_rec) {
    read(audio_fd, buf, 4);
  } else {
    read(audio_fd, buf, 2);
  }
  return(TRUE);
}

boolean
adin_oss_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_oss_read(SP16 *buf, int sampnum)
{
#ifndef HAS_OSS
  return -2;
#else
  int size,cnt,i;
  audio_buf_info info;
  fd_set rfds;
  struct timeval tv;
  int status;

  /* check for incoming samples in device buffer */
  /* if there is at least one sample fragment, go next */
  /* if not exist, wait for the data to come for at most MAXPOLLINTERVAL msec */
  /* if no sample fragment has come in the MAXPOLLINTERVAL period, go next */
  FD_ZERO(&rfds);
  FD_SET(audio_fd, &rfds);
  tv.tv_sec = 0;
  tv.tv_usec = MAXPOLLINTERVAL * 1000;
  status = select(audio_fd+1, &rfds, NULL, NULL, &tv);
  if (status < 0) {
    /* select() failed */
    jlog("Error: adin_oss: failed to poll device\n");
    return(-2);                 /* error */
  }
  if (FD_ISSET(audio_fd, &rfds)) { /* has some data */
    /* get sample num that can be read without blocking */
    if (ioctl(audio_fd, SNDCTL_DSP_GETISPACE, &info) == -1) {
      jlog("Error: adin_oss: failed to get number of samples in the buffer\n");
      return(-2);
    }
    /* get them as much as possible */
    size = sampnum * sizeof(SP16);
    if (size > info.bytes) size = info.bytes;
    if (size < frag_size) size = frag_size;
    size &= ~ 1;                /* Force 16bit alignment */
    cnt = read(audio_fd, buf, size);
    if ( cnt < 0 ) {
      jlog("Error: adin_oss: failed to read samples\n");
      return ( -2 );
    }
    cnt /= sizeof(short);

    if (stereo_rec) {
      /* remove R channel */
      for(i=1;i<cnt;i+=2) buf[(i-1)/2]=buf[i];
      cnt/=2;
    }
    
    if (need_swap) swap_sample_bytes(buf, cnt);
  } else {                      /* no data after waiting */
    jlog("Warning: adin_oss: no data fragment after %d msec?\n", MAXPOLLINTERVAL);
    cnt = 0;
  }

  return(cnt);
#endif /* HAS_OSS */
}

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