コード例 #1
0
// open & setup audio device
// return: 1=success 0=fail
static int init(int rate, int channels, int format, int flags) {

  int smpwidth, smpfmt;
  int rv = AL_DEFAULT_OUTPUT;

  smpfmt = fmt2sgial(&format, &smpwidth);

  mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_InitInfo, rate, (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));

  { /* from /usr/share/src/dmedia/audio/setrate.c */

    double frate, realrate;
    ALpv x[2];

    if(ao_subdevice) {
      rv = alGetResourceByName(AL_SYSTEM, ao_subdevice, AL_OUTPUT_DEVICE_TYPE);
      if (!rv) {
	mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InvalidDevice);
	return 0;
      }
    }

    frate = rate;

    x[0].param = AL_RATE;
    x[0].value.ll = alDoubleToFixed(rate);
    x[1].param = AL_MASTER_CLOCK;
    x[1].value.i = AL_CRYSTAL_MCLK_TYPE;

    if (alSetParams(rv,x, 2)<0) {
      mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantSetParms_Samplerate, alGetErrorString(oserror()));
    }

    if (x[0].sizeOut < 0) {
      mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantSetAlRate);
    }

    if (alGetParams(rv,x, 1)<0) {
      mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantGetParms, alGetErrorString(oserror()));
    }

    realrate = alFixedToDouble(x[0].value.ll);
    if (frate != realrate) {
      mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_SampleRateInfo, realrate, frate);
    }
    sample_rate = (int)realrate;
  }

  bytes_per_frame = channels * smpwidth;

  ao_data.samplerate = sample_rate;
  ao_data.channels = channels;
  ao_data.format = format;
  ao_data.bps = sample_rate * bytes_per_frame;
  ao_data.buffersize=131072;
  ao_data.outburst = ao_data.buffersize/16;

  ao_config = alNewConfig();

  if (!ao_config) {
    mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitConfigError, alGetErrorString(oserror()));
    return 0;
  }

  if(alSetChannels(ao_config, channels) < 0 ||
     alSetWidth(ao_config, smpwidth) < 0 ||
     alSetSampFmt(ao_config, smpfmt) < 0 ||
     alSetQueueSize(ao_config, sample_rate) < 0 ||
     alSetDevice(ao_config, rv) < 0) {
    mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitConfigError, alGetErrorString(oserror()));
    return 0;
  }

  ao_port = alOpenPort("mplayer", "w", ao_config);

  if (!ao_port) {
    mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitOpenAudioFailed, alGetErrorString(oserror()));
    return 0;
  }

  // printf("ao_sgi, init: port %d config %d\n", ao_port, ao_config);
  queue_size = alGetQueueSize(ao_config);
  return 1;

}
コード例 #2
0
ファイル: ao_sgi.c プロジェクト: HermiG/mplayer2
// open & setup audio device
// return: 1=success 0=fail
static int init(int rate, int channels, int format, int flags) {

  int smpwidth, smpfmt;
  int rv = AL_DEFAULT_OUTPUT;

  smpfmt = fmt2sgial(&format, &smpwidth);

  mp_tmsg(MSGT_AO, MSGL_INFO, "[AO SGI] init: Samplerate: %iHz Channels: %s Format %s\n", rate, (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));

  { /* from /usr/share/src/dmedia/audio/setrate.c */

    double frate, realrate;
    ALpv x[2];

    if(ao_subdevice) {
      rv = alGetResourceByName(AL_SYSTEM, ao_subdevice, AL_OUTPUT_DEVICE_TYPE);
      if (!rv) {
	mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SGI] play: invalid device.\n");
	return 0;
      }
    }

    frate = rate;

    x[0].param = AL_RATE;
    x[0].value.ll = alDoubleToFixed(rate);
    x[1].param = AL_MASTER_CLOCK;
    x[1].value.i = AL_CRYSTAL_MCLK_TYPE;

    if (alSetParams(rv,x, 2)<0) {
      mp_tmsg(MSGT_AO, MSGL_WARN, "[AO SGI] init: setparams failed: %s\nCould not set desired samplerate.\n", alGetErrorString(oserror()));
    }

    if (x[0].sizeOut < 0) {
      mp_tmsg(MSGT_AO, MSGL_WARN, "[AO SGI] init: AL_RATE was not accepted on the given resource.\n");
    }

    if (alGetParams(rv,x, 1)<0) {
      mp_tmsg(MSGT_AO, MSGL_WARN, "[AO SGI] init: getparams failed: %s\n", alGetErrorString(oserror()));
    }

    realrate = alFixedToDouble(x[0].value.ll);
    if (frate != realrate) {
      mp_tmsg(MSGT_AO, MSGL_INFO, "[AO SGI] init: samplerate is now %f (desired rate is %f)\n", realrate, frate);
    }
    sample_rate = (int)realrate;
  }

  bytes_per_frame = channels * smpwidth;

  ao_data.samplerate = sample_rate;
  ao_data.channels = channels;
  ao_data.format = format;
  ao_data.bps = sample_rate * bytes_per_frame;
  ao_data.buffersize=131072;
  ao_data.outburst = ao_data.buffersize/16;

  ao_config = alNewConfig();

  if (!ao_config) {
    mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SGI] init: %s\n", alGetErrorString(oserror()));
    return 0;
  }

  if(alSetChannels(ao_config, channels) < 0 ||
     alSetWidth(ao_config, smpwidth) < 0 ||
     alSetSampFmt(ao_config, smpfmt) < 0 ||
     alSetQueueSize(ao_config, sample_rate) < 0 ||
     alSetDevice(ao_config, rv) < 0) {
    mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SGI] init: %s\n", alGetErrorString(oserror()));
    return 0;
  }

  ao_port = alOpenPort("mplayer", "w", ao_config);

  if (!ao_port) {
    mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SGI] init: Unable to open audio channel: %s\n", alGetErrorString(oserror()));
    return 0;
  }

  // printf("ao_sgi, init: port %d config %d\n", ao_port, ao_config);
  queue_size = alGetQueueSize(ao_config);
  return 1;

}