C++ snd_pcm_sw_params_set_start_threshold函数代码示例

static void *alsa_init(const char *device, unsigned rate, unsigned latency)
{
   alsa_t *alsa = (alsa_t*)calloc(1, sizeof(alsa_t));
   if (!alsa)
      return NULL;

   snd_pcm_hw_params_t *params = NULL;
   snd_pcm_sw_params_t *sw_params = NULL;

   unsigned latency_usec = latency * 1000;
   unsigned channels = 2;
   unsigned periods = 4;
   snd_pcm_format_t format;

   const char *alsa_dev = "default";
   if (device)
      alsa_dev = device;

   snd_pcm_uframes_t buffer_size;

   TRY_ALSA(snd_pcm_open(&alsa->pcm, alsa_dev, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK));
   TRY_ALSA(snd_pcm_hw_params_malloc(&params));
   alsa->has_float = find_float_format(alsa->pcm, params);
   format = alsa->has_float ? SND_PCM_FORMAT_FLOAT : SND_PCM_FORMAT_S16;

   TRY_ALSA(snd_pcm_hw_params_any(alsa->pcm, params));
   TRY_ALSA(snd_pcm_hw_params_set_access(alsa->pcm, params, SND_PCM_ACCESS_RW_INTERLEAVED));
   TRY_ALSA(snd_pcm_hw_params_set_format(alsa->pcm, params, format));
   TRY_ALSA(snd_pcm_hw_params_set_channels(alsa->pcm, params, channels));
   TRY_ALSA(snd_pcm_hw_params_set_rate(alsa->pcm, params, rate, 0));

   TRY_ALSA(snd_pcm_hw_params_set_buffer_time_near(alsa->pcm, params, &latency_usec, NULL));
   TRY_ALSA(snd_pcm_hw_params_set_periods_near(alsa->pcm, params, &periods, NULL));

   TRY_ALSA(snd_pcm_hw_params(alsa->pcm, params));

   snd_pcm_hw_params_get_period_size(params, &buffer_size, NULL);
   RARCH_LOG("ALSA: Period size: %d frames\n", (int)buffer_size);
   snd_pcm_hw_params_get_buffer_size(params, &buffer_size);
   RARCH_LOG("ALSA: Buffer size: %d frames\n", (int)buffer_size);
   alsa->buffer_size = snd_pcm_frames_to_bytes(alsa->pcm, buffer_size);
   alsa->can_pause = snd_pcm_hw_params_can_pause(params);
   RARCH_LOG("ALSA: Can pause: %s.\n", alsa->can_pause ? "yes" : "no");

   TRY_ALSA(snd_pcm_sw_params_malloc(&sw_params));
   TRY_ALSA(snd_pcm_sw_params_current(alsa->pcm, sw_params));
   TRY_ALSA(snd_pcm_sw_params_set_start_threshold(alsa->pcm, sw_params, buffer_size / 2));
   TRY_ALSA(snd_pcm_sw_params(alsa->pcm, sw_params));

   snd_pcm_hw_params_free(params);
   snd_pcm_sw_params_free(sw_params);

   return alsa;

error:
   RARCH_ERR("ALSA: Failed to initialize...\n");
   if (params)
      snd_pcm_hw_params_free(params);

   if (sw_params)
      snd_pcm_sw_params_free(sw_params);

   if (alsa)
   {
      if (alsa->pcm)
         snd_pcm_close(alsa->pcm);

      free(alsa);
   }
   return NULL;
}

static void
alsa_configure (struct sound_device *sd)
{
  int val, err, dir;
  unsigned uval;
  struct alsa_params *p = (struct alsa_params *) sd->data;
  snd_pcm_uframes_t buffer_size;

  xassert (p->handle != 0);

  err = snd_pcm_hw_params_malloc (&p->hwparams);
  if (err < 0)
    alsa_sound_perror ("Could not allocate hardware parameter structure", err);

  err = snd_pcm_sw_params_malloc (&p->swparams);
  if (err < 0)
    alsa_sound_perror ("Could not allocate software parameter structure", err);

  err = snd_pcm_hw_params_any (p->handle, p->hwparams);
  if (err < 0)
    alsa_sound_perror ("Could not initialize hardware parameter structure", err);

  err = snd_pcm_hw_params_set_access (p->handle, p->hwparams,
                                      SND_PCM_ACCESS_RW_INTERLEAVED);
  if (err < 0)
    alsa_sound_perror ("Could not set access type", err);

  val = sd->format;
  err = snd_pcm_hw_params_set_format (p->handle, p->hwparams, val);
  if (err < 0)
    alsa_sound_perror ("Could not set sound format", err);

  uval = sd->sample_rate;
  err = snd_pcm_hw_params_set_rate_near (p->handle, p->hwparams, &uval, 0);
  if (err < 0)
    alsa_sound_perror ("Could not set sample rate", err);

  val = sd->channels;
  err = snd_pcm_hw_params_set_channels (p->handle, p->hwparams, val);
  if (err < 0)
    alsa_sound_perror ("Could not set channel count", err);

  err = snd_pcm_hw_params (p->handle, p->hwparams);
  if (err < 0)
    alsa_sound_perror ("Could not set parameters", err);


  err = snd_pcm_hw_params_get_period_size (p->hwparams, &p->period_size, &dir);
  if (err < 0)
    alsa_sound_perror ("Unable to get period size for playback", err);

  err = snd_pcm_hw_params_get_buffer_size (p->hwparams, &buffer_size);
  if (err < 0)
    alsa_sound_perror ("Unable to get buffer size for playback", err);

  err = snd_pcm_sw_params_current (p->handle, p->swparams);
  if (err < 0)
    alsa_sound_perror ("Unable to determine current swparams for playback",
                       err);

  /* Start the transfer when the buffer is almost full */
  err = snd_pcm_sw_params_set_start_threshold (p->handle, p->swparams,
                                               (buffer_size / p->period_size)
                                               * p->period_size);
  if (err < 0)
    alsa_sound_perror ("Unable to set start threshold mode for playback", err);

  /* Allow the transfer when at least period_size samples can be processed */
  err = snd_pcm_sw_params_set_avail_min (p->handle, p->swparams, p->period_size);
  if (err < 0)
    alsa_sound_perror ("Unable to set avail min for playback", err);

  err = snd_pcm_sw_params (p->handle, p->swparams);
  if (err < 0)
    alsa_sound_perror ("Unable to set sw params for playback\n", err);

  snd_pcm_hw_params_free (p->hwparams);
  p->hwparams = NULL;
  snd_pcm_sw_params_free (p->swparams);
  p->swparams = NULL;

  err = snd_pcm_prepare (p->handle);
  if (err < 0)
    alsa_sound_perror ("Could not prepare audio interface for use", err);

  if (sd->volume > 0)
    {
      int chn;
      snd_mixer_t *handle;
      snd_mixer_elem_t *e;
      const char *file = sd->file ? sd->file : DEFAULT_ALSA_SOUND_DEVICE;

      if (snd_mixer_open (&handle, 0) >= 0)
        {
          if (snd_mixer_attach (handle, file) >= 0
              && snd_mixer_load (handle) >= 0
              && snd_mixer_selem_register (handle, NULL, NULL) >= 0)
            for (e = snd_mixer_first_elem (handle);
                 e;
                 e = snd_mixer_elem_next (e))
//.........这里部分代码省略.........

Error AudioDriverALSA::init() {

	active=false;
	thread_exited=false;
	exit_thread=false;
	pcm_open = false;
	samples_in = NULL;
	samples_out = NULL;

	mix_rate = GLOBAL_DEF("audio/mix_rate",44100);
	output_format = OUTPUT_STEREO;
	channels = 2;


	int                  status;
	snd_pcm_hw_params_t *hwparams;
	snd_pcm_sw_params_t *swparams;

#define CHECK_FAIL(m_cond)\
	if (m_cond) {\
		fprintf(stderr,"ALSA ERR: %s\n",snd_strerror(status));\
		snd_pcm_close(pcm_handle);\
		ERR_FAIL_COND_V(m_cond,ERR_CANT_OPEN);\
	}

	//todo, add
	//6 chans - "plug:surround51"
	//4 chans - "plug:surround40";

	status = snd_pcm_open(&pcm_handle, "default", SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);

	ERR_FAIL_COND_V( status<0, ERR_CANT_OPEN );

	snd_pcm_hw_params_alloca(&hwparams);

	status = snd_pcm_hw_params_any(pcm_handle, hwparams);
	CHECK_FAIL( status<0 );

	status = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
	CHECK_FAIL( status<0 );

	//not interested in anything else
	status = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE);
	CHECK_FAIL( status<0 );

	//todo: support 4 and 6
	status = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, 2);
	CHECK_FAIL( status<0 );

	status = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &mix_rate, NULL);
	CHECK_FAIL( status<0 );

	int latency = GLOBAL_DEF("audio/output_latency",25);
	buffer_size = nearest_power_of_2( latency * mix_rate / 1000 );

	// set buffer size from project settings
	status = snd_pcm_hw_params_set_buffer_size_near(pcm_handle, hwparams, &buffer_size);
	CHECK_FAIL( status<0 );

	// make period size 1/8 
	period_size = buffer_size >> 3;
	status = snd_pcm_hw_params_set_period_size_near(pcm_handle, hwparams, &period_size, NULL);
	CHECK_FAIL( status<0 );

	unsigned int periods=2;
	status = snd_pcm_hw_params_set_periods_near(pcm_handle, hwparams, &periods, NULL);
	CHECK_FAIL( status<0 );

	status = snd_pcm_hw_params(pcm_handle,hwparams);
	CHECK_FAIL( status<0 );

	//snd_pcm_hw_params_free(&hwparams);


	snd_pcm_sw_params_alloca(&swparams);

	status = snd_pcm_sw_params_current(pcm_handle, swparams);
	CHECK_FAIL( status<0 );

	status = snd_pcm_sw_params_set_avail_min(pcm_handle, swparams, period_size);
	CHECK_FAIL( status<0 );

	status = snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams, 1);
	CHECK_FAIL( status<0 );

	status = snd_pcm_sw_params(pcm_handle, swparams);
	CHECK_FAIL( status<0 );

	samples_in = memnew_arr(int32_t, period_size*channels);
	samples_out = memnew_arr(int16_t, period_size*channels);

	snd_pcm_nonblock(pcm_handle, 0);

	mutex=Mutex::create();
	thread = Thread::create(AudioDriverALSA::thread_func, this);

	return OK;
};

/* setup alsa data transfer behavior */
static inline int alsa_set_swparams(ao_device *device)
{
	ao_alsa_internal *internal  = (ao_alsa_internal *) device->internal;
	snd_pcm_sw_params_t *params;
        snd_pcm_uframes_t boundary;
	int err;

	/* allocate the software parameter structure */
	snd_pcm_sw_params_alloca(&params);

	/* fetch the current software parameters */
	err = snd_pcm_sw_params_current(internal->pcm_handle, params);
	if (err < 0){
          adebug("snd_pcm_sw_params_current() failed.\n");
          return err;
        }

#if 0 /* the below causes more trouble than it cures */
	/* allow transfers to start when there is one period */
	err = snd_pcm_sw_params_set_start_threshold(internal->pcm_handle,
                                                    params, internal->period_size);
	if (err < 0){
          adebug("snd_pcm_sw_params_set_start_threshold() failed.\n");
          //return err;
        }

	/* require a minimum of one full transfer in the buffer */
	err = snd_pcm_sw_params_set_avail_min(internal->pcm_handle, params,
                                              internal->period_size);
	if (err < 0){
          adebug("snd_pcm_sw_params_set_avail_min() failed.\n");
          //return err;
        }
#endif

	/* do not align transfers; this is obsolete/deprecated in ALSA
           1.x where the transfer alignemnt is always 1 (except for
           buggy drivers like VIA 82xx which still demand aligned
           transfers regardless of setting, in violation of the ALSA
           API docs) */
	err = snd_pcm_sw_params_set_xfer_align(internal->pcm_handle, params, 1);
	if (err < 0){
          adebug("snd_pcm_sw_params_set_xfer_align() failed.\n");
          //return err;
        }


        /* get the boundary size */
        err = snd_pcm_sw_params_get_boundary(params,&boundary);
	if (err < 0){
          adebug("snd_pcm_sw_params_get_boundary() failed.\n");
        }else{

          /* force a work-ahead silence buffer; this is a fix, again for
             VIA 82xx, where non-MMIO transfers will buffer into
             period-size transfers, but the last transfer is usually
             undersized and playback falls off the end of the submitted
             data. */
          err = snd_pcm_sw_params_set_silence_size(internal->pcm_handle, params, boundary);
          if (err < 0){
            adebug("snd_pcm_sw_params_set_silence_size() failed.\n");
            //return err;
          }
        }

	/* commit the params structure to ALSA */
	err = snd_pcm_sw_params(internal->pcm_handle, params);
	if (err < 0){
          adebug("snd_pcm_sw_params() failed.\n");
          return err;
        }

	return 1;
}

//.........这里部分代码省略.........
        			case 24: pcm_format_id = SND_PCM_FORMAT_U24_3BE; break;
        			case 32: pcm_format_id = SND_PCM_FORMAT_U32_BE;  break;
        		}
        		break;

        	case BLT_PCM_SAMPLE_FORMAT_FLOAT_BE:
                ATX_LOG_FINE("sample format is BLT_PCM_SAMPLE_FORMAT_FLOAT_LE");
        		switch (format->bits_per_sample) {
        			case 32: pcm_format_id = SND_PCM_FORMAT_FLOAT_BE; break;
        		}
        		break;
        }

        if (pcm_format_id == SND_PCM_FORMAT_UNKNOWN) {
            return BLT_ERROR_INVALID_MEDIA_TYPE;
        }
        ior = snd_pcm_hw_params_set_format(self->device_handle, 
                                           hw_params,
                                           pcm_format_id);
        if (ior != 0) {
            ATX_LOG_WARNING_2("snd_pcm_hw_params_set_format() failed (%d:%s)", ior, snd_strerror(ior));
            return BLT_FAILURE;
        }

        /* set the period size */
        ior = snd_pcm_hw_params_set_period_size_near(self->device_handle, 
                                                     hw_params,
                                                     &period_size,
                                                     NULL);
        if (ior != 0) {
            ATX_LOG_WARNING_2("snd_pcm_hw_params_set_period_size_near() failed (%d:%s)", ior, snd_strerror(ior));
            return BLT_FAILURE;
        }
        
                                                
        /* set the buffer time (duration) */
        ior = snd_pcm_hw_params_set_buffer_time_near(self->device_handle,
                                                     hw_params, 
                                                     &buffer_time,
                                                     NULL);
        if (ior != 0) {
            ATX_LOG_WARNING_2("snd_pcm_hw_params_set_buffer_time_near() failed (%d:%s)", ior, snd_strerror(ior));
            return BLT_FAILURE;
        }

        /* get the actual buffer size */
        snd_pcm_hw_params_get_buffer_size(hw_params, &buffer_size);

        /* activate this configuration */
        ior = snd_pcm_hw_params(self->device_handle, hw_params);
        if (ior != 0) {
            ATX_LOG_WARNING_2("snd_pcm_hw_params() failed (%d:%s)", ior, snd_strerror(ior));
            return BLT_FAILURE;
        }

        /* configure the software parameters */
        snd_pcm_sw_params_alloca_no_assert(&sw_params);
        snd_pcm_sw_params_current(self->device_handle, sw_params);

        /* set the start threshold to 1/2 the buffer size */
        snd_pcm_sw_params_set_start_threshold(self->device_handle, 
                                              sw_params, 
                                              buffer_size/2);

        /* set the buffer alignment */
        /* NOTE: this call is now obsolete */
        /* snd_pcm_sw_params_set_xfer_align(self->device_handle, sw_params, 1); */

        /* activate the sofware parameters */
        ior = snd_pcm_sw_params(self->device_handle, sw_params);
        if (ior != 0) {
            ATX_LOG_SEVERE_2("snd_pcm_sw_params() failed (%d:%s)", ior, snd_strerror(ior));
            return BLT_FAILURE;
        }

        /* print status info */
        {
            snd_pcm_uframes_t val;
            ATX_LOG_FINER_1("sample type = %x", pcm_format_id);
            if (rate != format->sample_rate) {
                ATX_LOG_FINER_1("actual sample = %d", rate);
            }
            ATX_LOG_FINER_1("actual buffer time = %d", (int)buffer_time);
            ATX_LOG_FINER_1("buffer size = %d", (int)buffer_size); 
            snd_pcm_sw_params_get_start_threshold(sw_params, &val);
            ATX_LOG_FINER_1("start threshold = %d", (int)val); 
            snd_pcm_sw_params_get_stop_threshold(sw_params, &val);
            ATX_LOG_FINER_1("stop threshold = %d", (int)val); 
            snd_pcm_hw_params_get_period_size(hw_params, &val, NULL);
            ATX_LOG_FINER_1("period size = %d", (int)val);
        }

        break;
    }

    /* update the state */
    AlsaOutput_SetState(self, BLT_ALSA_OUTPUT_STATE_CONFIGURED);

    return BLT_SUCCESS;
}

static int
set_swparams (GstAlsaSrc * alsa)
{
  int err;
  snd_pcm_sw_params_t *params;

  snd_pcm_sw_params_malloc (&params);

  /* get the current swparams */
  CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config);
  /* allow the transfer when at least period_size samples can be processed */
  CHECK (snd_pcm_sw_params_set_avail_min (alsa->handle, params,
          alsa->period_size), set_avail);
  /* start the transfer on first read */
  CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params,
          0), start_threshold);
  /* use monotonic timestamping */
  CHECK (snd_pcm_sw_params_set_tstamp_mode (alsa->handle, params,
          SND_PCM_TSTAMP_MMAP), tstamp_mode);

#if GST_CHECK_ALSA_VERSION(1,0,16)
  /* snd_pcm_sw_params_set_xfer_align() is deprecated, alignment is always 1 */
#else
  /* align all transfers to 1 sample */
  CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align);
#endif

  /* write the parameters to the recording device */
  CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params);

  snd_pcm_sw_params_free (params);
  return 0;

  /* ERRORS */
no_config:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
        ("Unable to determine current swparams for playback: %s",
            snd_strerror (err)));
    snd_pcm_sw_params_free (params);
    return err;
  }
start_threshold:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
        ("Unable to set start threshold mode for playback: %s",
            snd_strerror (err)));
    snd_pcm_sw_params_free (params);
    return err;
  }
set_avail:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
        ("Unable to set avail min for playback: %s", snd_strerror (err)));
    snd_pcm_sw_params_free (params);
    return err;
  }
tstamp_mode:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
        ("Unable to set tstamp mode for playback: %s", snd_strerror (err)));
    snd_pcm_sw_params_free (params);
    return err;
  }
#if !GST_CHECK_ALSA_VERSION(1,0,16)
set_align:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
        ("Unable to set transfer align for playback: %s", snd_strerror (err)));
    snd_pcm_sw_params_free (params);
    return err;
  }
#endif
set_sw_params:
  {
    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
        ("Unable to set sw params for playback: %s", snd_strerror (err)));
    snd_pcm_sw_params_free (params);
    return err;
  }
}

//.........这里部分代码省略.........
    if (rc < 0) {
        printf("Failed to set PCM device to mono: %s\n",
                snd_strerror(rc));
        exit(1);
    }
    /* 44100 bits/second sampling rate (CD quality) */
    val = rate;
    rc = snd_pcm_hw_params_set_rate_near(this->m_pHandle, this->m_pParams,
                    &val, 0);
    if (rc < 0) {
        printf("Failed to set PCM device to sample rate =%d: %s\n",
                rate,snd_strerror(rc));
        exit(1);
    }
	if(val != rate){
		printf("Rate doesn't match (request %iHz, get %iHz\n",rate,val);
		exit(1);
	}


	rc = snd_pcm_hw_params_set_buffer_time_near(this->m_pHandle, this->m_pParams, &buffer_time, &dir);
    if (rc < 0) {
        fprintf(stderr, "Failed to set PCM device to buffer time=%d: %s\n",
                    val,snd_strerror(rc));
        exit(1);
    }

	rc = snd_pcm_hw_params_set_period_time_near(this->m_pHandle, this->m_pParams, &period_time, &dir);
	if (rc < 0) {
        fprintf(stderr, "Failed to set PCM device to period time=%d: %s\n",
                    val,snd_strerror(rc));
        exit(1);
    }
	
	//chunk size
	rc = snd_pcm_hw_params_get_period_size(this->m_pParams, &period_size, &dir);
	if(rc < 0){
		fprintf(stderr, "Failed to get period size for capture: %s\n", snd_strerror(rc));
		exit(1);
	}
	this->m_frames = period_size;

	rc = snd_pcm_hw_params_get_buffer_size(this->m_pParams, &buffer_size);
	if(rc < 0){
		fprintf(stderr, "Failed to get buffer size for capture: %s\n", snd_strerror(rc));
		exit(1);
	}

#ifdef DEBUG
	printf("Recorder alsa driver hw params setting\n");
	printf("period size =%d frames\n", (int)period_size);
	printf("buffer size =%d frames\n", (int)buffer_size);
	snd_pcm_hw_params_get_period_time(this->m_pParams,&val, &dir);
	printf("period time =%d us\n",val);
	snd_pcm_hw_params_get_buffer_time(this->m_pParams,&val, &dir);
	printf("buffer time =%d us\n", val);
	snd_pcm_hw_params_get_periods(this->m_pParams, &val, &dir);
	printf("period per buffer =%d frames\n", val);
#endif	

   /* Write the parameters to the driver */
    rc = snd_pcm_hw_params(this->m_pHandle, this->m_pParams);
    if (rc < 0) {
        printf("unable to set hw parameters: %s\n",
                    snd_strerror(rc));
        exit(1);
    }
    
#ifdef ALSA_OPT
	snd_pcm_sw_params_t *swparams;
	snd_pcm_sw_params_alloca(&swparams);

    /* get the current swparams */
    rc = snd_pcm_sw_params_current(this->m_pHandle, swparams);
    if (rc < 0) {
            printf("Unable to determine current swparams for playback: %s\n", snd_strerror(rc));
            exit(1);
    }
    /* start the transfer when the buffer is almost full: */
    /* (buffer_size / avail_min) * avail_min */
    rc = snd_pcm_sw_params_set_start_threshold(this->m_pHandle, swparams, (buffer_size / period_size) * period_size);
    if (rc < 0) {
            printf("Unable to set start threshold mode for playback: %s\n", snd_strerror(rc));
            exit(1);
    }

    rc = snd_pcm_sw_params_set_avail_min(this->m_pHandle, swparams, period_size);
    if (rc < 0) {
            printf("Unable to set avail min for playback: %s\n", snd_strerror(rc));
            exit(1);
    }
    /* write the parameters to the playback device */
    rc = snd_pcm_sw_params(this->m_pHandle, swparams);
    if (rc < 0) {
            printf("Unable to set sw params for playback: %s\n", snd_strerror(rc));
            exit(1);
    }

#endif
}

//.........这里部分代码省略.........
	if (err < 0) {
		printf("Access type not available");
		exit(EXIT_FAILURE);
	}

    printf("format = %s, channels = %d, rate = %d\n",
            snd_pcm_format_name(hwparams.format),hwparams.channels,hwparams.rate);

	//err = snd_pcm_hw_params_set_format(handle, params, hwparams.format);
    err = snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE);
	if (err < 0) {
		printf("Sample format non available");
		exit(EXIT_FAILURE);
	}

	//err = snd_pcm_hw_params_set_channels(handle, params, hwparams.channels);
    err = snd_pcm_hw_params_set_channels(handle, params, 2);
	if (err < 0) {
		printf("Channels count non available");
		exit(EXIT_FAILURE);
	}

#if 0
	//add by yjc 2012/08/21
	err = set_audio_clk_freq(hwparams.rate);
	if (err < 0){
		printf("set_audio_clk_freq fail..........\n");
		exit(EXIT_FAILURE);
	}
#endif

	err = snd_pcm_hw_params_set_rate(handle, params, hwparams.rate, 0);
    if (err < 0) {
		printf("Rate non available");
		exit(EXIT_FAILURE);
	}


	err = snd_pcm_hw_params_get_buffer_time_max(params,
						    &buffer_time, 0);
	assert(err >= 0);
	if (buffer_time > 500000)
		buffer_time = 500000;


	period_time = buffer_time / 4;

	err = snd_pcm_hw_params_set_period_time_near(handle, params,
		                        &period_time, 0);
	assert(err >= 0);

	err = snd_pcm_hw_params_set_buffer_time_near(handle, params,
							     &buffer_time, 0);

	assert(err >= 0);

	err = snd_pcm_hw_params(handle, params);
	if (err < 0) {
		printf("Unable to install hw params:");
		exit(EXIT_FAILURE);
	}
	snd_pcm_hw_params_get_period_size(params, &chunk_size, 0);
	snd_pcm_hw_params_get_buffer_size(params, &buffer_size);
	if (chunk_size == buffer_size) {
		printf("Can't use period equal to buffer size (%lu == %lu)",
		      chunk_size, buffer_size);
		exit(EXIT_FAILURE);
	}

	snd_pcm_sw_params_current(handle, swparams);

	err = snd_pcm_sw_params_set_avail_min(handle, swparams, chunk_size);

	if(stream == SND_PCM_STREAM_PLAYBACK)
		start_threshold = (buffer_size / chunk_size) * chunk_size;
	else
		start_threshold = 1;
	err = snd_pcm_sw_params_set_start_threshold(handle, swparams, start_threshold);
	assert(err >= 0);

	err = snd_pcm_sw_params_set_stop_threshold(handle, swparams, buffer_size);
	assert(err >= 0);

	if (snd_pcm_sw_params(handle, swparams) < 0) {
		printf("unable to install sw params:");
		exit(EXIT_FAILURE);
	}

	//bits_per_sample = snd_pcm_format_physical_width(hwparams.format);
    bits_per_sample = snd_pcm_format_physical_width(SND_PCM_FORMAT_S16_LE);
	//bits_per_frame = bits_per_sample * hwparams.channels;
	bits_per_frame = bits_per_sample * 2;
	chunk_bytes = chunk_size * bits_per_frame / 8;

    printf("chunk_size = %d,chunk_bytes = %d,buffer_size = %d\n\n",
        (int)chunk_size,chunk_bytes,(int)buffer_size);

	return chunk_size;

}

//.........这里部分代码省略.........
    ui_error( UI_ERROR_ERROR, "couldn't set period size %d on '%s'.",
                              (int)sound_periodsize, pcm_name );
    snd_pcm_close( pcm_handle );
    init_running = 0;
    return 1;
  }

  if( first_init && ( exact_periodsize < sound_periodsize / 1.5 ||
		      exact_periodsize > sound_periodsize * 1.5    ) ) {
    fprintf( stderr,
              "The period size %d is not supported by your hardware. "
              "Using %d instead.\n", (int)sound_periodsize,
              (int)exact_periodsize );
  }

  periods = nperiods;
/* Set number of periods. Periods used to be called fragments. */
  if( snd_pcm_hw_params_set_periods_near( pcm_handle, hw_params, &periods,
                                          NULL ) < 0 ) {
    settings_current.sound = 0;
    ui_error( UI_ERROR_ERROR, "couldn't set periods on '%s'.", pcm_name );
    snd_pcm_close( pcm_handle );
    init_running = 0;
    return 1;
  }

  if( first_init && periods != nperiods ) {
    fprintf( stderr, "%d periods is not supported by your hardware. "
                    	     "Using %d instead.\n", nperiods, periods );
  }

  snd_pcm_hw_params_get_buffer_size( hw_params, &exact_bsize );

  /* Apply HW parameter settings to */
  /* PCM device and prepare device  */

  if( snd_pcm_hw_params( pcm_handle, hw_params ) < 0 ) {
    settings_current.sound = 0;
    ui_error( UI_ERROR_ERROR,"couldn't set hw_params on %s", pcm_name );
    snd_pcm_close( pcm_handle );
    init_running = 0;
    return 1;
  }

  snd_pcm_sw_params_alloca( &sw_params );
  if( ( err = snd_pcm_sw_params_current( pcm_handle, sw_params ) ) < 0 ) {
    ui_error( UI_ERROR_ERROR,"couldn't get sw_params from %s: %s", pcm_name,
              snd_strerror ( err ) );
    snd_pcm_close( pcm_handle );
    init_running = 0;
    return 1;
  }

  if( ( err = snd_pcm_sw_params_set_start_threshold( pcm_handle,
		     sw_params, exact_periodsize * ( nperiods - 1 ) ) ) < 0 ) {
    ui_error( UI_ERROR_ERROR,"couldn't set start_treshold on %s: %s", pcm_name,
              snd_strerror ( err ) );
    snd_pcm_close( pcm_handle );
    init_running = 0;
    return 1;
  }

  if( !avail_min )
    avail_min = exact_periodsize >> 1;
  if( snd_pcm_sw_params_set_avail_min( pcm_handle,
		    sw_params, avail_min ) < 0 ) {
#if SND_LIB_VERSION < 0x10010
    if( ( err = snd_pcm_sw_params_set_sleep_min( pcm_handle,
    		    sw_params, 1 ) ) < 0 ) {
	fprintf( stderr, "Unable to set minimal sleep 1 for %s: %s\n", pcm_name,
              snd_strerror ( err ) );
    }
#else
    fprintf( stderr, "Unable to set avail min %s: %s\n", pcm_name,
    	     snd_strerror( err ) );
#endif
  }

#if SND_LIB_VERSION < 0x10010
  if( ( err = snd_pcm_sw_params_set_xfer_align( pcm_handle, sw_params, 1 ) ) < 0 ) {
    ui_error( UI_ERROR_ERROR,"couldn't set xfer_allign on %s: %s", pcm_name,
              snd_strerror ( err ) );
    init_running = 0;
    return 1;
  }
#endif
  
  if( ( err = snd_pcm_sw_params( pcm_handle, sw_params ) ) < 0 ) {
    ui_error( UI_ERROR_ERROR,"couldn't set sw_params on %s: %s", pcm_name,
              snd_strerror ( err ) );
    init_running = 0;
    return 1;
  }

  if( first_init ) snd_output_stdio_attach(&output, stdout, 0);

  first_init = 0;
  init_running = 0;
  return 0;	/* success */
}

//.........这里部分代码省略.........
        }
        qLog(QAudioInput) << "channels = " << vval;

        snd_pcm_hw_params_get_rate(hwparams, &vval, &dir);
        if ( (int)vval != (int)frequency ) {
            qLog(QAudioInput) << QString("frequency type not set, want %1 got %2").arg(frequency).arg(vval);
            frequency = vval;
        }
        qLog(QAudioInput) << "rate =" <<  vval << " bps";

        snd_pcm_hw_params_get_period_time(hwparams,&period_time, &dir);
        qLog(QAudioInput) << "period time =" << period_time << " us";
        snd_pcm_hw_params_get_period_size(hwparams,&period_size, &dir);
        qLog(QAudioInput) << "period size =" << (int)period_size;
        snd_pcm_hw_params_get_buffer_time(hwparams,&buffer_time, &dir);
        qLog(QAudioInput) << "buffer time =" << buffer_time;
        snd_pcm_hw_params_get_buffer_size(hwparams,(snd_pcm_uframes_t *) &buffer_size);
        qLog(QAudioInput) << "buffer size =" << (int)buffer_size;
        snd_pcm_hw_params_get_periods(hwparams, &vval, &dir);
        qLog(QAudioInput) << "periods per buffer =" << vval;
        snd_pcm_hw_params_get_rate_numden(hwparams, &vval, &vval2);
        qLog(QAudioInput) << QString("exact rate = %1/%2 bps").arg(vval).arg(vval2).toLatin1().constData();
        vval = snd_pcm_hw_params_get_sbits(hwparams);
        qLog(QAudioInput) << "significant bits =" << vval;
        snd_pcm_hw_params_get_tick_time(hwparams,&vval, &dir);
        qLog(QAudioInput) << "tick time =" << vval;
        vval = snd_pcm_hw_params_is_batch(hwparams);
        qLog(QAudioInput) << "is batch =" << vval;
        vval = snd_pcm_hw_params_is_block_transfer(hwparams);
        qLog(QAudioInput) << "is block transfer =" << vval;
        vval = snd_pcm_hw_params_is_double(hwparams);
        qLog(QAudioInput) << "is double =" << vval;
        vval = snd_pcm_hw_params_is_half_duplex(hwparams);
        qLog(QAudioInput) << "is half duplex =" << vval;
        vval = snd_pcm_hw_params_is_joint_duplex(hwparams);
        qLog(QAudioInput) << "is joint duplex =" << vval;
        vval = snd_pcm_hw_params_can_overrange(hwparams);
        qLog(QAudioInput) << "can overrange =" << vval;
        vval = snd_pcm_hw_params_can_mmap_sample_resolution(hwparams);
        qLog(QAudioInput) << "can mmap =" << vval;
        vval = snd_pcm_hw_params_can_pause(hwparams);
        qLog(QAudioInput) << "can pause =" << vval;
        vval = snd_pcm_hw_params_can_resume(hwparams);
        qLog(QAudioInput) << "can resume =" << vval;
        vval = snd_pcm_hw_params_can_sync_start(hwparams);
        qLog(QAudioInput) << "can sync start =" << vval;

        snd_pcm_sw_params_t *swparams;
        snd_pcm_sw_params_alloca(&swparams);
        err = snd_pcm_sw_params_current(handle, swparams);
        if ( err < 0 ) {
            qWarning( "QAudioInput: snd_pcm_sw_params_current: err %d",err);
        }
        err = snd_pcm_sw_params_set_start_threshold(handle,swparams,period_size);
        if ( err < 0 ) {
            qWarning( "QAudioInput: snd_pcm_sw_params_set_start_threshold: err %d",err);
        }
        err = snd_pcm_sw_params_set_avail_min(handle, swparams,period_size);
        if ( err < 0 ) {
            qWarning( "QAudioInput: snd_pcm_sw_params_set_avail_min: err %d",err);
        }
        err = snd_pcm_sw_params_set_xfer_align(handle, swparams, 1);
        if ( err < 0 ) {
            qWarning( "QAudioInput: snd_pcm_sw_params_set_xfer_align: err %d",err);
        }
        err = snd_pcm_sw_params(handle, swparams);
        if ( err < 0 ) {
            qWarning( "QAudioInput: snd_pcm_sw_params: err %d",err);
        }

        snd_pcm_prepare(handle);
        snd_pcm_start(handle);

        int     count = snd_pcm_poll_descriptors_count(handle);
        pollfd  *pfds = new pollfd[count];

        snd_pcm_poll_descriptors(handle, pfds, count);

        for (int i = 0; i < count; ++i)
        {
            if ((pfds[i].events & POLLIN) != 0) {
                notifier = new QSocketNotifier(pfds[i].fd, QSocketNotifier::Read);
                QObject::connect(notifier,
                                 SIGNAL(activated(int)),
                                 input,
                                 SIGNAL(readyRead()));

                break;
            }
        }

        if (notifier == NULL) {
            rc = false;
        }

        delete pfds;
    }

    return rc;
}

static int sndo_pcm_initialize(sndo_pcm_t *pcm)
{
	int err;
	snd_pcm_uframes_t boundary;
	snd_pcm_uframes_t p_period_size = ~0UL, c_period_size = ~0UL;
	snd_pcm_uframes_t p_buffer_size = ~0UL, c_buffer_size = ~0UL;

	if (pcm->playback) {
		err = snd_pcm_hw_params(pcm->playback, pcm->p_hw_params);
		if (err < 0)
			return err;
		err = snd_pcm_hw_params_get_period_size(pcm->p_hw_params, &p_period_size, NULL);
		if (err < 0)
			return err;
		err = snd_pcm_hw_params_get_buffer_size(pcm->p_hw_params, &p_buffer_size);
		if (err < 0)
			return err;
	}
	if (pcm->capture) {
		err = snd_pcm_hw_params(pcm->capture, pcm->c_hw_params);
		if (err < 0)
			return err;
		err = snd_pcm_hw_params_get_period_size(pcm->c_hw_params, &c_period_size, NULL);
		if (err < 0)
			return err;
		err = snd_pcm_hw_params_get_buffer_size(pcm->c_hw_params, &c_buffer_size);
		if (err < 0)
			return err;
	}
	if (p_period_size < c_period_size)
		pcm->transfer_block = p_period_size;
	else
		pcm->transfer_block = c_period_size;
	if (p_buffer_size < c_buffer_size)
		pcm->ring_size = p_buffer_size;
	else
		pcm->ring_size = c_buffer_size;
	if (pcm->playback) {
		err = snd_pcm_sw_params_get_boundary(pcm->p_sw_params, &boundary);
		if (err < 0)
			return err;
		err = snd_pcm_sw_params_set_start_threshold(pcm->playback, pcm->p_sw_params, boundary);
		if (err < 0)
			return err;
		err = snd_pcm_sw_params_set_stop_threshold(pcm->playback, pcm->p_sw_params, pcm->xrun == SNDO_PCM_XRUN_IGNORE ? boundary : pcm->ring_size);
		if (err < 0)
			return err;
		err = snd_pcm_sw_params_set_xfer_align(pcm->playback, pcm->p_sw_params, 1);
		if (err < 0)
			return err;
		err = snd_pcm_sw_params_set_avail_min(pcm->playback, pcm->p_sw_params, pcm->transfer_block);
		if (err < 0)
			return err;
		err = snd_pcm_sw_params(pcm->playback, pcm->p_sw_params);
		if (err < 0)
			return err;
	}
	if (pcm->capture) {
		err = snd_pcm_sw_params_get_boundary(pcm->c_sw_params, &boundary);
		if (err < 0)
			return err;
		err = snd_pcm_sw_params_set_start_threshold(pcm->capture, pcm->c_sw_params, boundary);
		if (err < 0)
			return err;
		err = snd_pcm_sw_params_set_stop_threshold(pcm->capture, pcm->c_sw_params, pcm->xrun == SNDO_PCM_XRUN_IGNORE ? boundary : pcm->ring_size);
		if (err < 0)
			return err;
		err = snd_pcm_sw_params_set_xfer_align(pcm->capture, pcm->c_sw_params, 1);
		if (err < 0)
			return err;
		err = snd_pcm_sw_params_set_avail_min(pcm->capture, pcm->c_sw_params, pcm->transfer_block);
		if (err < 0)
			return err;
		err = snd_pcm_sw_params(pcm->capture, pcm->c_sw_params);
		if (err < 0)
			return err;
	}
	pcm->initialized = 1;
	return 0;
}

static int pcm_open(pcm_handle_t* pcm, const pcm_desc_t* desc)
{
  const snd_pcm_format_t fmt = SND_PCM_FORMAT_S16_LE;
  snd_pcm_stream_t stm;
  int err;

  if (desc->flags & PCM_FLAG_IN) stm = SND_PCM_STREAM_CAPTURE;
  else stm = SND_PCM_STREAM_PLAYBACK;

  err = snd_pcm_open
    (&pcm->pcm, desc->name, stm, SND_PCM_NONBLOCK);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_0);

  err = snd_pcm_hw_params_malloc(&pcm->hw_params);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_1);

  err = snd_pcm_hw_params_any(pcm->pcm, pcm->hw_params);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_2);

  err = snd_pcm_hw_params_set_access
    (pcm->pcm, pcm->hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_2);

  err = snd_pcm_hw_params_set_format(pcm->pcm, pcm->hw_params, fmt);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_2);

  err = snd_pcm_hw_params_set_rate
    (pcm->pcm, pcm->hw_params, desc->fsampl, 0);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_2);

  pcm->nchan = desc->nchan;
  pcm->wchan = (size_t)snd_pcm_format_physical_width(fmt) / 8;
  pcm->scale = pcm->nchan * pcm->wchan;

  err = snd_pcm_hw_params_set_channels
    (pcm->pcm, pcm->hw_params, desc->nchan);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_2);
  
  err = snd_pcm_hw_params(pcm->pcm, pcm->hw_params);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_2);

  err = snd_pcm_sw_params_malloc(&pcm->sw_params);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_2);

  err = snd_pcm_sw_params_current(pcm->pcm, pcm->sw_params);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_3);

#if 1
  err = snd_pcm_sw_params_set_avail_min
    (pcm->pcm, pcm->sw_params, 1024);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_3);
#endif

#if 1
  err = snd_pcm_sw_params_set_start_threshold
    (pcm->pcm, pcm->sw_params, 0U);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_3);
#endif

  err = snd_pcm_sw_params(pcm->pcm, pcm->sw_params);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_3);
  
  err = snd_pcm_prepare(pcm->pcm);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_3);

  pcm->rpos = 0;
  pcm->wpos = 0;
  pcm->nsampl = (size_t)desc->fsampl * 10;
  pcm->buf = malloc(pcm->nsampl * pcm->scale);
  if (pcm->buf == NULL) goto on_error_3;

  return 0;

 on_error_3:
  snd_pcm_sw_params_free(pcm->sw_params);
 on_error_2:
  snd_pcm_hw_params_free(pcm->hw_params);
 on_error_1:
  snd_pcm_close(pcm->pcm);
 on_error_0:
  return -1;
  
}

//.........这里部分代码省略.........
    /* Set up the information for recording from the audio channelport.  This
     will include data specified by the user, such as data in
     aTIesrFAInstance->encoding, etc.   If setup fails, then the audio channel
     should be shut down and failure reported. */

    /* Setup hardware parameters of the ALSA pcm */
    rtnStatus = snd_pcm_hw_params_malloc( &hw_params );
    if( rtnStatus < 0 )
        goto openExit;

    rtnStatus = snd_pcm_hw_params_any( ALSAData->alsa_handle, hw_params );
    if( rtnStatus < 0 )
        goto openExit;

    /* There will only be a single channel */
    rtnStatus = snd_pcm_hw_params_set_channels( ALSAData->alsa_handle,
            hw_params, 1 );
    if( rtnStatus < 0 )
        goto openExit;

    /* Even though only one channel, must specify type of read. */
    rtnStatus = snd_pcm_hw_params_set_access( ALSAData->alsa_handle,
            hw_params,
            SND_PCM_ACCESS_RW_INTERLEAVED );
    if( rtnStatus < 0 )
        goto openExit;


    /* Format type - only 16 bit linear for now */
    rtnStatus = snd_pcm_hw_params_set_format( ALSAData->alsa_handle,
            hw_params,
            SND_PCM_FORMAT_S16 );
    if( rtnStatus < 0 )
        goto openExit;

    /* Set sample rate, it must be exactly supported */
    rtnStatus = snd_pcm_hw_params_set_rate( ALSAData->alsa_handle,
            hw_params,
            aTIesrFAInstance->sample_rate, 0 );
    if( rtnStatus < 0 )
        goto openExit;


    /* Size of ALSA PCM ring buffer is set to approximately the desired 
     number of multiples of read_samples in the buffer */
    numSamples = aTIesrFAInstance->num_audio_buffer_frames *
            ALSAData->read_samples;
    rtnStatus = snd_pcm_hw_params_set_buffer_size_near( ALSAData->alsa_handle,
            hw_params, &numSamples );
    if( rtnStatus < 0 )
        goto openExit;


    /* Set the hardware parameters in the PCM*/
    rtnStatus = snd_pcm_hw_params( ALSAData->alsa_handle, hw_params );
    if( rtnStatus < 0 )
        goto openExit;



    /* Set software parameters to interrupt at the end of an audio buffer 
     number of samples and to start-up manually */
    rtnStatus = snd_pcm_sw_params_malloc( &sw_params );
    if( rtnStatus < 0 )
        goto openExit;

    rtnStatus = snd_pcm_sw_params_current( ALSAData->alsa_handle, sw_params );
    if( rtnStatus < 0 )
        goto openExit;

    /* Number of samples needed in PCM buffer prior to interrupt */
    rtnStatus = snd_pcm_sw_params_set_avail_min( ALSAData->alsa_handle,
            sw_params, ALSAData->read_samples );
    if( rtnStatus < 0 )
        goto openExit;

    /* Set start threshold so startup is done manually */
    rtnStatus = snd_pcm_sw_params_set_start_threshold( ALSAData->alsa_handle,
            sw_params, ULONG_MAX );
    if( rtnStatus < 0 )
        goto openExit;

    /* Channel opened successfully */
    openSuccess = 1;


openExit:

    if( hw_params != NULL )
        snd_pcm_hw_params_free( hw_params );

    if( sw_params != NULL )
        snd_pcm_sw_params_free( sw_params );


    if( openSuccess )
        return TIesrFA_ALSAErrNone;

    return TIesrFA_ALSAErrFail;
}

int setparams_set(snd_pcm_t *handle,
		  snd_pcm_hw_params_t *params,
		  snd_pcm_sw_params_t *swparams,
		  const char *id)
{
	int err;
	snd_pcm_uframes_t val;
	unsigned int sleep_min = 0;

	err = snd_pcm_hw_params(handle, params);
	if (err < 0) {
		printf("Unable to set hw params for %s: %s\n", id, snd_strerror(err));
		return err;
	}
	err = snd_pcm_sw_params_current(handle, swparams);
	if (err < 0) {
		printf("Unable to determine current swparams for %s: %s\n", id, snd_strerror(err));
		return err;
	}
	err = snd_pcm_sw_params_set_start_threshold(handle, swparams, 0x7fffffff);
	if (err < 0) {
		printf("Unable to set start threshold mode for %s: %s\n", id, snd_strerror(err));
		return err;
	}
	tick_time_ok = 0;
	if (tick_time > 0) {
		int time, ttime;
		snd_pcm_hw_params_get_period_time(params, &time, NULL);
		 snd_pcm_hw_params_get_tick_time(params, &ttime, NULL);
		if (time < ttime) {
			printf("Skipping to set minimal sleep: period time < tick time\n");
		} else if (ttime <= 0) {
			printf("Skipping to set minimal sleep: tick time <= 0 (%i)\n", ttime);
		} else {
			sleep_min = tick_time / ttime;
			if (sleep_min <= 0)
				sleep_min = 1;
			err = snd_pcm_sw_params_set_sleep_min(handle, swparams, sleep_min);
			if (err < 0) {
				printf("Unable to set minimal sleep %i for %s: %s\n", sleep_min, id, snd_strerror(err));
				return err;
			}
			tick_time_ok = sleep_min * ttime;
		}
	}
	if (!block)
		val = 4;
	else
		snd_pcm_hw_params_get_period_size(params, &val, NULL);
	if (tick_time_ok > 0)
		val = 16;
	err = snd_pcm_sw_params_set_avail_min(handle, swparams, val);
	if (err < 0) {
		printf("Unable to set avail min for %s: %s\n", id, snd_strerror(err));
		return err;
	}
	val = !block ? 4 : 1;
	err = snd_pcm_sw_params_set_xfer_align(handle, swparams, val);
	if (err < 0) {
		printf("Unable to set transfer align for %s: %s\n", id, snd_strerror(err));
		return err;
	}
	err = snd_pcm_sw_params(handle, swparams);
	if (err < 0) {
		printf("Unable to set sw params for %s: %s\n", id, snd_strerror(err));
		return err;
	}
	return 0;
}

//.........这里部分代码省略.........
	{
		oss << "cannot set access type : " << snd_strerror (err) << endl;
		throw oss.str();
	}

	/* Set Sample format: Signed 16bit little endian. */
	if ((err = snd_pcm_hw_params_set_format (playback_handle, hw_params, SND_PCM_FORMAT_S16_LE)) < 0)
	{
		oss << "cannot set sample format : " << snd_strerror (err) << endl;
		throw oss.str();
	}

	/* Set the Sample rate. */
	if ((err = snd_pcm_hw_params_set_rate (playback_handle, hw_params, 22050, 0)) < 0)
	{
		oss << "cannot set sample rate : " << snd_strerror (err) << endl;
		throw oss.str();
	}

	/* Set Channel number (MONO). */
	if ((err = snd_pcm_hw_params_set_channels (playback_handle, hw_params, 1)) < 0)
	{
		oss << "cannot set channel count : " << snd_strerror (err) << endl;
		throw oss.str();
	}


	if ((err = snd_pcm_hw_params_set_buffer_size(playback_handle, hw_params, 2048)) < 0)
	{
		oss << "cannot set channel buffer size : " << snd_strerror (err) << endl;
		throw oss.str();
	}



	/* Apply these parameters. */
	if ((err = snd_pcm_hw_params (playback_handle, hw_params)) < 0)
	{
		oss << "cannot apply parameters : " << snd_strerror (err) << endl;
		throw oss.str();
	}

	snd_pcm_uframes_t bufferSize;
	snd_pcm_hw_params_get_buffer_size( hw_params, &bufferSize );
	//cout << "initAlsa: Buffer size = " << bufferSize << " frames." << endl;

	/* Free memoray allocated for snd_pcm_hw_params_t */
	snd_pcm_hw_params_free (hw_params);

	/* tell ALSA to wake us up whenever 4096 or more frames
	   of playback data can be delivered. Also, tell
	   ALSA that we'll start the device ourselves.
	*/

	/* Allocate snd_pcm_sw_params_t structure. */
	if ((err = snd_pcm_sw_params_malloc (&sw_params)) < 0)
	{
		oss << "cannot allocate software parameters structure : " << snd_strerror (err) << endl;
		throw oss.str();
	}

	/* Get the current software configuration*/
	if ((err = snd_pcm_sw_params_current (playback_handle, sw_params)) < 0)
	{
		oss << "cannot initialize software parameters structure : " << snd_strerror (err) << endl;
		throw oss.str();
	}

	/* Set the wake up point to 2048 (92.9 ms). The minimum data available before asking*/
	/* for new ones. */
	if ((err = snd_pcm_sw_params_set_avail_min (playback_handle, sw_params, 2048U)) < 0)
	{
		oss << "cannot set minimum available count : " << snd_strerror (err) << endl;
		throw oss.str();
	}

	/* Set when ALSA starts to play. */
	if ((err = snd_pcm_sw_params_set_start_threshold (playback_handle, sw_params, 1024U)) < 0)
	{
		oss << "cannot set start mode : " << snd_strerror (err) << endl;
		throw oss.str();
	}

	/* Apply parameters. */
	if ((err = snd_pcm_sw_params (playback_handle, sw_params)) < 0)
	{
		oss << "cannot apply software parameters : " << snd_strerror (err) << endl;
		throw oss.str();
	}

	/* the interface will interrupt the kernel every 4096 frames, and ALSA
	   will wake up this program very soon after that.
	*/

	if ((err = snd_pcm_prepare (playback_handle)) < 0)
	{
		oss << "cannot prepare audio interface for use : " << snd_strerror (err) << endl;
		throw oss.str();
	}
}