adin_mic_linux_alsa.c

julius version 4.12.about sound recognition.

/**
 * @file   adin_mic_linux_alsa.c
 *
 * <JA>
 * @brief  マイク掐蜗 (Linux/ALSA)
 *
 * ALSA API を蝗脱する·マイク掐蜗のための你レベル簇眶ですˉ
 * 蝗脱には ALSA サウンドドライバ〖がインスト〖ルされていることが涩妥ですˉ
 *
 * サウンドカ〖ドが 16bit モノラル で峡不できることが涩寇ですˉ
 *
 * JuliusはLinuxではミキサ〖デバイスの肋年を办磊乖いませんˉ峡不デバイスの
 * 联买∈マイク/ライン∷や峡不ボリュ〖ムの拇泪は alsamixer など戮のツ〖ルで
 * 乖なって布さいˉ
 *
 * 剩眶サウンドカ〖ドはサポ〖トされていませんˉ剩眶のサウンドカ〖ドが
 * インスト〖ルされている眷圭·呵介の１つが脱いられますˉ
 *
 * バ〖ジョン 1.x に滦炳していますˉ1.0.13 で瓢侯を澄千しましたˉ
 *
 * デバイス叹は "default" が蝗脱されますˉ茨董恃眶 ALSADEV で恃构できますˉ
 * </JA>
 * <EN>
 * @brief  Microphone input on Linux/ALSA
 *
 * Low level I/O functions for microphone input on Linux using
 * Advanced Linux Sound Architechture (ALSA) API, developed on version 0.9.x.
 *
 * Julius does not alter any mixer device setting at all on Linux.  You should
 * configure the mixer for recording source (mic/line) and recording volume
 * correctly using other audio tool such as alsamixer.
 *
 * Note that sound card should support 16bit monaural recording, and multiple
 * cards are not supported (in that case the first one will be used).
 *
 * This file supports alsa version 1.x, and tested on 1.0.13.
 *
 * The default PCM device name is "default", and can be overwritten by
 * environment variable "ALSADEV".
 * </EN>
 *
 * @sa http://www.alsa-project.org/
 *
 * @author Akinobu LEE
 * @date   Sun Feb 13 16:18:26 2005
 *
 * $Revision: 1.9 $
 * 
 */
/*
 * 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 <fcntl.h>

#ifdef HAS_ALSA

#if defined(HAVE_ALSA_ASOUNDLIB_H)
#include <alsa/asoundlib.h>
#elif defined(HAVE_SYS_ASOUNDLIB_H)
#include <sys/asoundlib.h>
#endif

static snd_pcm_t *handle;	///< Audio handler
static char *pcm_name = "default"; ///< Name of the PCM device, can be overridden by env ALSADEV
static int latency = 32;	///< Lantency time in msec.  You can override this value by specifying environment valuable "LATENCY_MSEC".
static boolean need_swap;	///< Whether samples need byte swap

#if (SND_LIB_MAJOR == 0)
static struct pollfd *ufds;	///< Poll descriptor
static int count;		///< Poll descriptor count
#endif

#define MAXPOLLINTERVAL 300	///< Read timeout in msec.

#endif /* HAS_ALSA */

#ifdef HAS_ALSA
/** 
 * Output detailed device information.
 * 
 * @param pcm_name [in] device name string
 * @param handle [in] pcm audio handler
 * 
 */
static void
output_card_info(char *pcm_name, snd_pcm_t *handle)
{
  int err;
  snd_ctl_t *ctl;
  snd_ctl_card_info_t *info;
  snd_pcm_info_t *pcminfo;
  snd_ctl_card_info_alloca(&info);
  snd_pcm_info_alloca(&pcminfo);
  char ctlname[30];
  int card;
  
  /* get PCM information to set current device and subdevice name */
  if ((err = snd_pcm_info(handle, pcminfo)) < 0) {
    jlog("Warning: adin_alsa: failed to obtain pcm info\n");
    jlog("Warning: adin_alsa: skip output of detailed audio device info\n");
    return;
  }
  /* open control associated with the pcm device name */
  card = snd_pcm_info_get_card(pcminfo);
  if (card < 0) {
    strcpy(ctlname, "default");
  } else {
    snprintf(ctlname, 30, "hw:%d", card);
  }
  if ((err = snd_ctl_open(&ctl, ctlname, 0)) < 0) {
    jlog("Warning: adin_alsa: failed to open control device \"%s\", \n", ctlname);
    jlog("Warning: adin_alsa: skip output of detailed audio device info\n");
    return;
  }
  /* get its card info */
  if ((err = snd_ctl_card_info(ctl, info)) < 0) {
    jlog("Warning: adin_alsa: unable to get card info for %s\n", ctlname);
    jlog("Warning: adin_alsa: skip output of detailed audio device info\n");
    snd_ctl_close(ctl);
    return;
  }

  /* get detailed PCM information of current device from control */
  if ((err = snd_ctl_pcm_info(ctl, pcminfo)) < 0) {
    jlog("Error: adin_alsa: unable to get pcm info from card control\n");
    jlog("Warning: adin_alsa: skip output of detailed audio device info\n");
    snd_ctl_close(ctl);
    return;
  }
  /* output */
  jlog("Stat: \"%s\": %s [%s] device %s [%s] %s\n",
       pcm_name,
       snd_ctl_card_info_get_id(info),
       snd_ctl_card_info_get_name(info),
       snd_pcm_info_get_id(pcminfo),
       snd_pcm_info_get_name(pcminfo),
       snd_pcm_info_get_subdevice_name(pcminfo));

  /* close controller */
  snd_ctl_close(ctl);

}
#endif /* HAS_ALSA */

/** 
 * Device initialization: check device capability and open for recording.
 * 
 * @param sfreq [in] required sampling frequency.
 * @param dummy [in] a dummy data
 * 
 * @return TRUE on success, FALSE on failure.
 */
boolean
adin_alsa_standby(int sfreq, void *dummy)
{
#ifndef HAS_ALSA
  jlog("Error: ALSA not compiled in\n");
  return FALSE;
#else
  int err;
  snd_pcm_hw_params_t *hwparams; ///< Pointer to device hardware parameters
#if (SND_LIB_MAJOR == 0)
  int actual_rate;		/* sample rate returned by hardware */
#else
  unsigned int actual_rate;		/* sample rate returned by hardware */
#endif
  int dir = 0;			/* comparison result of exact rate and given rate */
  char *p;

  /* check $ALSADEV for device name */
  if ((p = getenv("ALSADEV")) != NULL) {
    pcm_name = p;
    jlog("Stat: adin_alsa: device name from ALSADEV: \"%s\"\n", pcm_name);
  }

  /* open device in non-block mode) */
  if ((err = snd_pcm_open(&handle, pcm_name, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK)) < 0) {
    jlog("Error: adin_alsa: cannot open PCM device \"%s\" (%s)\n", pcm_name, snd_strerror(err));
    return(FALSE);
  }
  /* set device to non-block mode */
  if ((err = snd_pcm_nonblock(handle, 1)) < 0) {
    jlog("Error: adin_alsa: cannot set PCM device to non-blocking mode\n");
    return(FALSE);
  }

  /* allocate hwparam structure */
  snd_pcm_hw_params_alloca(&hwparams);

  /* initialize hwparam structure */
  if ((err = snd_pcm_hw_params_any(handle, hwparams)) < 0) {
    jlog("Error: adin_alsa: cannot initialize PCM device parameter structure (%s)\n", snd_strerror(err));
    return(FALSE);
  }

  /* set interleaved read/write format */
  if ((err = snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
    jlog("Error: adin_alsa: cannot set PCM device access mode (%s)\n", snd_strerror(err));
    return(FALSE);
  }

  /* set sample format */
#ifdef WORDS_BIGENDIAN
  /* try big endian, then little endian with byte swap */
  if ((err = snd_pcm_hw_params_set_format(handle, hwparams, SND_PCM_FORMAT_S16_BE)) >= 0) {
    need_swap = FALSE;
  } else if ((err = snd_pcm_hw_params_set_format(handle, hwparams, SND_PCM_FORMAT_S16_LE)) >= 0) {
    need_swap = TRUE;
  } else {
    jlog("Error: adin_alsa: cannot set PCM device format to signed 16bit (%s)\n", snd_strerror(err));
    return(FALSE);
  }
#else  /* LITTLE ENDIAN */
  /* try little endian, then big endian with byte swap */
  if ((err = snd_pcm_hw_params_set_format(handle, hwparams, SND_PCM_FORMAT_S16_LE)) >= 0) {
    need_swap = FALSE;
  } else if ((err = snd_pcm_hw_params_set_format(handle, hwparams, SND_PCM_FORMAT_S16_BE)) >= 0) {
    need_swap = TRUE;
  } else {
    jlog("Error: adin_alsa: cannot set PCM device format to signed 16bit (%s)\n", snd_strerror(err));
    return(FALSE);
  }
#endif
  /* set number of channels */
  if ((err = snd_pcm_hw_params_set_channels(handle, hwparams, 1)) < 0) {
    jlog("Error: adin_alsa: cannot set PCM channel to %d (%s)\n", 1, snd_strerror(err));
    return(FALSE);
  }
  
  /* set sample rate (if the exact rate is not supported by the hardware, use nearest possible rate */
#if (SND_LIB_MAJOR == 0)
  actual_rate = snd_pcm_hw_params_set_rate_near(handle, hwparams, sfreq, &dir);
  if (actual_rate < 0) {
    jlog("Error: adin_alsa: cannot set PCM device sample rate to %d (%s)\n", sfreq, snd_strerror(actual_rate));
    return(FALSE);
  }
#else
  actual_rate = sfreq;
  err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &actual_rate, &dir);
  if (err < 0) {
    jlog("Error: adin_alsa: cannot set PCM device sample rate to %d (%s)\n", sfreq, snd_strerror(err));
    return(FALSE);
  }
#endif
  if (actual_rate != sfreq) {
    jlog("Warning: adin_alsa: the exact rate %d Hz is not available by your PCM hardware.\n", sfreq);
    jlog("Warning: adin_alsa: using %d Hz instead.\n", actual_rate);
  }
  jlog("Stat: capture audio at %dHz\n", actual_rate);

  /* set period size */
  {
#if (SND_LIB_MAJOR == 0)
    int periodsize;		/* period size (bytes) */
    int actual_size;
    int maxsize, minsize;
#else
    unsigned int period_time, period_time_current;
    snd_pcm_uframes_t chunk_size;
    boolean has_current_period;
#endif
    boolean force = FALSE;
    
    /* set apropriate period size */
    if ((p = getenv("LATENCY_MSEC")) != NULL) {
      latency = atoi(p);
      jlog("Stat: adin_alsa: trying to set latency to %d msec from LATENCY_MSEC)\n", latency);
      force = TRUE;
    }

    /* get hardware max/min size */
#if (SND_LIB_MAJOR == 0)
    if ((maxsize = snd_pcm_hw_params_get_period_size_max(hwparams, &dir)) < 0) {
      jlog("Error: adin_alsa: cannot get maximum period size\n");