/**************************************************************************
 * 		      wodPlayer_ProcessMessages			[internal]
 */
static void wodPlayer_ProcessMessages(WINE_WAVEDEV* wwo)
{
    LPWAVEHDR           lpWaveHdr;
    enum win_wm_message	msg;
    DWORD_PTR		param;
    HANDLE		ev;
    int                 err;

    while (ALSA_RetrieveRingMessage(&wwo->msgRing, &msg, &param, &ev)) {
     TRACE("Received %s %lx\n", ALSA_getCmdString(msg), param);

	switch (msg) {
	case WINE_WM_PAUSING:
	    if ( snd_pcm_state(wwo->pcm) == SND_PCM_STATE_RUNNING )
	     {
                if ( snd_pcm_hw_params_can_pause(wwo->hw_params) )
		{
		    err = snd_pcm_pause(wwo->pcm, 1);
		    if ( err < 0 )
			ERR("pcm_pause failed: %s\n", snd_strerror(err));
		    wwo->state = WINE_WS_PAUSED;
		}
		else
		{
		    wodPlayer_Reset(wwo,FALSE);
		}
	     }
	    SetEvent(ev);
	    break;
	case WINE_WM_RESTARTING:
            if (wwo->state == WINE_WS_PAUSED)
            {
		if ( snd_pcm_state(wwo->pcm) == SND_PCM_STATE_PAUSED )
		 {
		    err = snd_pcm_pause(wwo->pcm, 0);
		    if ( err < 0 )
		        ERR("pcm_pause failed: %s\n", snd_strerror(err));
		 }
                wwo->state = WINE_WS_PLAYING;
            }
	    SetEvent(ev);
	    break;
	case WINE_WM_HEADER:
	    lpWaveHdr = (LPWAVEHDR)param;

	    /* insert buffer at the end of queue */
	    {
		LPWAVEHDR*	wh;
		for (wh = &(wwo->lpQueuePtr); *wh; wh = &((*wh)->lpNext));
		*wh = lpWaveHdr;
	    }
            if (!wwo->lpPlayPtr)
                wodPlayer_BeginWaveHdr(wwo,lpWaveHdr);
	    if (wwo->state == WINE_WS_STOPPED)
		wwo->state = WINE_WS_PLAYING;
	    break;
	case WINE_WM_RESETTING:
	    wodPlayer_Reset(wwo,TRUE);
	    SetEvent(ev);
	    break;
        case WINE_WM_BREAKLOOP:
            if (wwo->state == WINE_WS_PLAYING && wwo->lpLoopPtr != NULL) {
                /* ensure exit at end of current loop */
                wwo->dwLoops = 1;
            }
	    SetEvent(ev);
            break;
	case WINE_WM_CLOSING:
	    /* sanity check: this should not happen since the device must have been reset before */
	    if (wwo->lpQueuePtr || wwo->lpPlayPtr) ERR("out of sync\n");
	    wwo->hThread = 0;
	    wwo->state = WINE_WS_CLOSED;
	    SetEvent(ev);
	    ExitThread(0);
	    /* shouldn't go here */
	default:
	    FIXME("unknown message %d\n", msg);
	    break;
	}
    }
}

/**
 * Set up the snd_pcm_t object which was opened by the caller.  Set up
 * the configured settings and the audio format.
 */
static bool
alsa_setup(struct alsa_data *ad, struct audio_format *audio_format,
	   GError **error)
{
	snd_pcm_hw_params_t *hwparams;
	snd_pcm_sw_params_t *swparams;
	unsigned int sample_rate = audio_format->sample_rate;
	unsigned int channels = audio_format->channels;
	snd_pcm_uframes_t alsa_buffer_size;
	snd_pcm_uframes_t alsa_period_size;
	int err;
	const char *cmd = NULL;
	int retry = MPD_ALSA_RETRY_NR;
	unsigned int period_time, period_time_ro;
	unsigned int buffer_time;

	period_time_ro = period_time = ad->period_time;
configure_hw:
	/* configure HW params */
	snd_pcm_hw_params_alloca(&hwparams);
	cmd = "snd_pcm_hw_params_any";
	err = snd_pcm_hw_params_any(ad->pcm, hwparams);
	if (err < 0)
		goto error;

	if (ad->use_mmap) {
		err = snd_pcm_hw_params_set_access(ad->pcm, hwparams,
						   SND_PCM_ACCESS_MMAP_INTERLEAVED);
		if (err < 0) {
			g_warning("Cannot set mmap'ed mode on ALSA device \"%s\":  %s\n",
				  alsa_device(ad), snd_strerror(-err));
			g_warning("Falling back to direct write mode\n");
			ad->use_mmap = false;
		} else
			ad->writei = snd_pcm_mmap_writei;
	}

	if (!ad->use_mmap) {
		cmd = "snd_pcm_hw_params_set_access";
		err = snd_pcm_hw_params_set_access(ad->pcm, hwparams,
						   SND_PCM_ACCESS_RW_INTERLEAVED);
		if (err < 0)
			goto error;
		ad->writei = snd_pcm_writei;
	}

	err = alsa_output_setup_format(ad->pcm, hwparams, audio_format);
	if (err < 0) {
		g_set_error(error, alsa_output_quark(), err,
			    "ALSA device \"%s\" does not support format %s: %s",
			    alsa_device(ad),
			    sample_format_to_string(audio_format->format),
			    snd_strerror(-err));
		return false;
	}

	err = snd_pcm_hw_params_set_channels_near(ad->pcm, hwparams,
						  &channels);
	if (err < 0) {
		g_set_error(error, alsa_output_quark(), err,
			    "ALSA device \"%s\" does not support %i channels: %s",
			    alsa_device(ad), (int)audio_format->channels,
			    snd_strerror(-err));
		return false;
	}
	audio_format->channels = (int8_t)channels;

	err = snd_pcm_hw_params_set_rate_near(ad->pcm, hwparams,
					      &sample_rate, NULL);
	if (err < 0 || sample_rate == 0) {
		g_set_error(error, alsa_output_quark(), err,
			    "ALSA device \"%s\" does not support %u Hz audio",
			    alsa_device(ad), audio_format->sample_rate);
		return false;
	}
	audio_format->sample_rate = sample_rate;

	snd_pcm_uframes_t buffer_size_min, buffer_size_max;
	snd_pcm_hw_params_get_buffer_size_min(hwparams, &buffer_size_min);
	snd_pcm_hw_params_get_buffer_size_max(hwparams, &buffer_size_max);
	unsigned buffer_time_min, buffer_time_max;
	snd_pcm_hw_params_get_buffer_time_min(hwparams, &buffer_time_min, 0);
	snd_pcm_hw_params_get_buffer_time_max(hwparams, &buffer_time_max, 0);
	g_debug("buffer: size=%u..%u time=%u..%u",
		(unsigned)buffer_size_min, (unsigned)buffer_size_max,
		buffer_time_min, buffer_time_max);

	snd_pcm_uframes_t period_size_min, period_size_max;
	snd_pcm_hw_params_get_period_size_min(hwparams, &period_size_min, 0);
	snd_pcm_hw_params_get_period_size_max(hwparams, &period_size_max, 0);
	unsigned period_time_min, period_time_max;
	snd_pcm_hw_params_get_period_time_min(hwparams, &period_time_min, 0);
	snd_pcm_hw_params_get_period_time_max(hwparams, &period_time_max, 0);
	g_debug("period: size=%u..%u time=%u..%u",
		(unsigned)period_size_min, (unsigned)period_size_max,
		period_time_min, period_time_max);
//.........这里部分代码省略.........

void* DAUDIO_Open(INT32 mixerIndex, INT32 deviceID, int isSource,
                  int encoding, float sampleRate, int sampleSizeInBits,
                  int frameSize, int channels,
                  int isSigned, int isBigEndian, int bufferSizeInBytes) {
    snd_pcm_format_mask_t* formatMask;
    snd_pcm_format_t format;
    int dir;
    int ret = 0;
    AlsaPcmInfo* info = NULL;
    /* snd_pcm_uframes_t is 64 bit on 64-bit systems */
    snd_pcm_uframes_t alsaPeriodSize = 0;
    snd_pcm_uframes_t alsaBufferSizeInFrames = 0;


    TRACE0("> DAUDIO_Open\n");
#ifdef USE_TRACE
    // for using ALSA debug dump methods
    if (ALSA_OUTPUT == NULL) {
        snd_output_stdio_attach(&ALSA_OUTPUT, stdout, 0);
    }
#endif

    info = (AlsaPcmInfo*) malloc(sizeof(AlsaPcmInfo));
    if (!info) {
        ERROR0("Out of memory\n");
        return NULL;
    }
    memset(info, 0, sizeof(AlsaPcmInfo));

    ret = openPCMfromDeviceID(deviceID, &(info->handle), isSource, FALSE /* do open device*/);
    if (ret == 0) {
        // set to blocking mode
        snd_pcm_nonblock(info->handle, 0);
        ret = snd_pcm_hw_params_malloc(&(info->hwParams));
        if (ret != 0) {
            ERROR1("  snd_pcm_hw_params_malloc returned error %d\n", ret);
        } else {
            ret = -1;
            if (getAlsaFormatFromFormat(&format, frameSize / channels, sampleSizeInBits,
                                        isSigned, isBigEndian, encoding)) {
                if (setHWParams(info,
                                sampleRate,
                                channels,
                                bufferSizeInBytes / frameSize,
                                format)) {
                    info->frameSize = frameSize;
#ifdef ALSA_PCM_NEW_HW_PARAMS_API
                    ret = snd_pcm_hw_params_get_period_size(info->hwParams, &alsaPeriodSize, &dir);
                    info->periodSize = (int) alsaPeriodSize;
                    if (ret < 0) {
                        ERROR1("ERROR: snd_pcm_hw_params_get_period: %s\n", snd_strerror(ret));
                    }
                    snd_pcm_hw_params_get_periods(info->hwParams, &(info->periods), &dir);
                    snd_pcm_hw_params_get_buffer_size(info->hwParams, &alsaBufferSizeInFrames);
                    info->bufferSizeInBytes = (int) alsaBufferSizeInFrames * frameSize;
#else
                    info->periodSize = snd_pcm_hw_params_get_period_size(info->hwParams, &dir);
                    info->periods = snd_pcm_hw_params_get_periods(info->hwParams, &dir);
                    info->bufferSizeInBytes = snd_pcm_hw_params_get_buffer_size(info->hwParams) * frameSize;
                    ret = 0;
#endif
                    TRACE3("  DAUDIO_Open: period size = %d frames, periods = %d. Buffer size: %d bytes.\n",
                           (int) info->periodSize, info->periods, info->bufferSizeInBytes);
                }
            }
        }
        if (ret == 0) {
            // set software parameters
            ret = snd_pcm_sw_params_malloc(&(info->swParams));
            if (ret != 0) {
                ERROR1("snd_pcm_hw_params_malloc returned error %d\n", ret);
            } else {
                if (!setSWParams(info)) {
                    ret = -1;
                }
            }
        }
        if (ret == 0) {
            // prepare device
            ret = snd_pcm_prepare(info->handle);
            if (ret < 0) {
                ERROR1("ERROR: snd_pcm_prepare: %s\n", snd_strerror(ret));
            }
        }

#ifdef GET_POSITION_METHOD2
        if (ret == 0) {
            ret = snd_pcm_status_malloc(&(info->positionStatus));
            if (ret != 0) {
                ERROR1("ERROR in snd_pcm_status_malloc: %s\n", snd_strerror(ret));
            }
        }
#endif
    }
    if (ret != 0) {
        DAUDIO_Close((void*) info, isSource);
        info = NULL;
    } else {
        // set to non-blocking mode
        snd_pcm_nonblock(info->handle, 1);
//.........这里部分代码省略.........

//音频采集线程
void * capture_audio_thread(void *para)
{
    int fdsound = 0;
    int readbyte = 0;


#if RECORD_CAPTURE_PCM
    FILE *fp = fopen("capture.pcm", "wb");
#endif

#if SILK_AUDIO_CODEC
    SKP_uint8 encode[MAX_BYTES_PER_FRAME * MAX_INPUT_FRAMES];
#endif

#if SILK_AUDIO_CODEC
    init_silk_encoder();//初始化silk codec
    printf("end of init silk encoder\n");
#endif














#if (SOUND_INTERFACE == SOUND_OSS)
    fdsound = open("/dev/dsp", O_RDONLY);
    if(fdsound<0)
    {
       perror("以只写方式打开音频设备");
       //return;
    }    

    printf("设置读音频设备参数 setup capture audio device parament\n");
    ioctl(fdsound, SNDCTL_DSP_SPEED, &Frequency);//采样频率
    ioctl(fdsound, SNDCTL_DSP_SETFMT, &format);//音频设备位宽
    ioctl(fdsound, SNDCTL_DSP_CHANNELS, &channels);//音频设备通道
    ioctl(fdsound, SNDCTL_DSP_SETFRAGMENT, &setting);//采样缓冲区

    while(flag_capture_audio)
    {
        if((readbyte=read(fdsound, pWriteHeader->buffer_capture, SAMPLERATE/1000*READMSFORONCE*sizeof(short))) < 0)
        {
            perror("读声卡数据");
        }
        else
        {
            //printf("readbyte=%d\n", readbyte);
#if RECORD_CAPTURE_PCM
            fwrite(pWriteHeader->buffer_capture, SAMPLERATE/1000*READMSFORONCE*sizeof(short), 1, fp);
#endif
            traceprintf("发送信号量 sem_capture\n");
            
            pWriteHeader->FrameNO = FrameNO++;

            time(&(pWriteHeader->time));
            //printf("cNO:%d, readbyte=%d,压缩后大小%d,pWriteHeader->vad=%d\n", pWriteHeader->FrameNO, readbyte, pWriteHeader->count_encode, pWriteHeader->vad);
            pthread_mutex_lock(&mutex_lock);
            n++;
            pWriteHeader->Valid = 1;
            pthread_mutex_unlock(&mutex_lock);
            pWriteHeader = pWriteHeader->pNext;
            sem_post(&sem_capture);
        }
    }
    close(fdsound);
#elif (SOUND_INTERFACE == SOUND_ALSA)
    printf("alsa xxxxxxxxxxxxxxx\n");
    struct timeval tv;
    struct timezone tz;
    /* Open PCM device for recording (capture). */
    rc = snd_pcm_open(&handle, "plughw:0,0", SND_PCM_STREAM_CAPTURE, 0);
    if (rc < 0) 
    {
        fprintf(stderr, "unable to open pcm device: %s\n", snd_strerror(rc));
        exit(1);
    }
    snd_pcm_hw_params_alloca(&params);/* Allocate a hardware parameters object. */    
    snd_pcm_hw_params_any(handle, params);/* Fill it in with default values. */

    /* Set the desired hardware parameters. */
    snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);/* Interleaved mode */
    snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE);/* Signed 16-bit little-endian format */
    snd_pcm_hw_params_set_channels(handle, params, CHANNELS);/* Two channels (stereo), On for mono */
    val = SAMPLERATE;
    snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir);/* SAMPLERATE bits/second sampling rate  */
    frames = SAMPLERATE/1000*READMSFORONCE;
    snd_pcm_hw_params_set_period_size_near(handle, params, &frames, &dir);/* Set period size to SAMPLES_FOR_EACH_TIME frames. */

    /* Write the parameters to the driver */
    rc = snd_pcm_hw_params(handle, params);
    if (rc < 0) 
//.........这里部分代码省略.........

int alsa_hook_hw_params(alsa_hook_t alsa_hook, snd_pcm_t *pcm, snd_pcm_hw_params_t *params)
{
	struct alsa_hook_stream_s *stream;

	snd_pcm_format_t format;
	snd_pcm_uframes_t period_size;
	snd_pcm_access_t access;
	int dir, ret;

	alsa_hook_get_stream(alsa_hook, pcm, &stream);
	if (unlikely((ret = alsa_hook_lock_write(alsa_hook, stream))))
		return ret;

	glc_log(alsa_hook->glc, GLC_DEBUG, "alsa_hook",
		 "%p: creating/updating configuration for stream %d",
		 stream->pcm, stream->id);

	/* extract information */
	if (unlikely((ret = snd_pcm_hw_params_get_format(params, &format)) < 0))
		goto err;
	stream->flags = 0; /* zero flags */
	stream->format = pcm_fmt_to_glc_fmt(format);
	if (unlikely(!stream->format)) {
		glc_log(alsa_hook->glc, GLC_ERROR, "alsa_hook",
			"%p: unsupported audio format 0x%02x", stream->pcm, format);
		ret = ENOTSUP;
		goto err;
	}
	if (unlikely((ret = snd_pcm_hw_params_get_rate(params, &stream->rate, &dir)) < 0))
		goto err;
	if (unlikely((ret = snd_pcm_hw_params_get_channels(params, &stream->channels)) < 0))
		goto err;
	if (unlikely((ret = snd_pcm_hw_params_get_period_size(params, &period_size, NULL)) < 0))
		goto err;
	if (unlikely((ret = snd_pcm_hw_params_get_access(params, &access)) < 0))
		goto err;
	if ((access == SND_PCM_ACCESS_RW_INTERLEAVED) ||
	    (access == SND_PCM_ACCESS_MMAP_INTERLEAVED))
		stream->flags |= GLC_AUDIO_INTERLEAVED;
	else if (access == SND_PCM_ACCESS_MMAP_COMPLEX) {
		stream->flags |= GLC_AUDIO_INTERLEAVED; /* convert to interleaved */
		stream->complex = 1; /* do conversion */
	} else {
		glc_log(alsa_hook->glc, GLC_ERROR, "alsa_hook",
			 "%p: unsupported access mode 0x%02x", stream->pcm, access);
		ret = ENOTSUP;
		goto err;
	}

	glc_log(alsa_hook->glc, GLC_DEBUG, "alsa_hook",
		 "%p: %d channels, rate %d, flags 0x%02x",
		 stream->pcm, stream->channels, stream->rate, stream->flags);

	stream->fmt = 1;
	if (alsa_hook->started) {
		if (unlikely((ret = alsa_hook_stream_init(alsa_hook, stream))))
			goto err;
	}

	alsa_hook_unlock_write(alsa_hook, stream);
	return 0;

err:
	glc_log(alsa_hook->glc, GLC_ERROR, "alsa_hook",
		 "%p: can't extract hardware configuration: %s (%d)",
		 stream->pcm, snd_strerror(ret), ret);

	alsa_hook_unlock_write(alsa_hook, stream);
	return ret;
}

int set_hwparams(snd_pcm_t *handle,
                        snd_pcm_hw_params_t *params,
                        snd_pcm_access_t access)
{
        unsigned int rrate;
        int err, dir;

	printf("set_hwparams\n");
        /* choose all parameters */
        err = snd_pcm_hw_params_any(handle, params);
        if (err < 0) {
                printf("Broken configuration for playback: no configurations available: %s\n", snd_strerror(err));
                return err;
        }
        /* set the interleaved read/write format */
        err = snd_pcm_hw_params_set_access(handle, params, access);
        if (err < 0) {
                printf("Access type not available for playback: %s\n", snd_strerror(err));
                return err;
        }
        /* set the sample format */
        err = snd_pcm_hw_params_set_format(handle, params, format);
        if (err < 0) {
                printf("Sample format not available for playback: %s\n", snd_strerror(err));
                return err;
        }
        /* set the count of channels */
        err = snd_pcm_hw_params_set_channels(handle, params, channels);
        if (err < 0) {
                printf("Channels count (%i) not available for playbacks: %s\n", channels, snd_strerror(err));
                return err;
        }
        /* set the stream rate */
        rrate = rate;
        err = snd_pcm_hw_params_set_rate_near(handle, params, &rrate, 0);
        if (err < 0) {
                printf("Rate %iHz not available for playback: %s\n", rate, snd_strerror(err));
                return err;
        }
        if (rrate != rate) {
                printf("Rate doesn't match (requested %iHz, get %iHz)\n", rate, err);
                return -EINVAL;
        }
	dir = 0;
        err = snd_pcm_hw_params_set_period_size_near(handle, params, &period_size, &dir);
        if (err < 0) {
                printf("Unable to set period size %i for playback: %s\n", (int)period_size, snd_strerror(err));
                return err;
        }
        err = snd_pcm_hw_params_set_periods(handle, params, periods, 0);
        if (err < 0) {
                printf("Unable to set periods %i for playback: %s\n", periods, snd_strerror(err));
                return err;
        }
        /* write the parameters to device */
        err = snd_pcm_hw_params(handle, params);
        if (err < 0) {
                printf("Unable to set hw params for playback: %s\n", snd_strerror(err));
                return err;
        }

        err = snd_pcm_hw_params_get_periods(params, &periods, &dir);
        if (err < 0) {
                printf("Unable to get periods for playback: %s\n", snd_strerror(err));
                return err;
        }
        err = snd_pcm_hw_params_get_period_size(params, &period_size, &dir);
        if (err < 0) {
                printf("Unable to get period size for playback: %s\n", snd_strerror(err));
                return err;
        }
        return 0;
}

/*******************************************************************************
 **
 ** Function        app_alsa_capture_loopback_open
 **
 ** Description     Open ALSA Capture channel (from loopback driver)
 **
 ** Parameters      p_open: Capture parameters
 **
 ** Returns         status
 **
 *******************************************************************************/
int app_alsa_capture_loopback_open(tAPP_ALSA_CAPTURE_OPEN *p_open)
{
    int mode = 0; /* Default is blocking */
    unsigned int nb_channels;
    int rv;
    snd_pcm_format_t format;
    snd_pcm_access_t access;

    APP_DEBUG0("Opening ALSA/Asound audio driver Capture");

    /* Sanity check if already opened */
    if (app_alsa_cb.p_capture_handle != NULL)
    {
        APP_DEBUG0("Capture was already opened");
    }

    /* check PCM Format parameter */
    format = app_alsa_get_pcm_format(p_open->format);
    if (format == SND_PCM_FORMAT_UNKNOWN)
    {
        return -1;
    }

    /* Check PCM access parameter */
    if (p_open->access == APP_ALSA_PCM_ACCESS_RW_INTERLEAVED)
    {
        access = SND_PCM_ACCESS_RW_INTERLEAVED;
    }
    else
    {
        APP_ERROR1("Unsupported PCM access:%d", p_open->access);
        return -1;
    }
    /* Check Blocking parameter */
    if (p_open->blocking == FALSE)
    {
        mode = SND_PCM_NONBLOCK;
    }

    /* check Stereo parameter */
    if (p_open->stereo)
    {
        nb_channels = 2;
    }
    else
    {
        nb_channels = 1;
    }

    /* Save the Capture open parameters */
    memcpy(&app_alsa_cb.capture_param, p_open, sizeof(app_alsa_cb.capture_param));

    /* Open ALSA driver */
    rv = snd_pcm_open(&app_alsa_cb.p_capture_handle, alsa_device_loopback,
            SND_PCM_STREAM_CAPTURE, mode);
    if (rv < 0)
    {
        APP_ERROR1("unable to open ALSA loopback device in Capture mode:%s", snd_strerror(rv));
        return rv;
    }
    APP_DEBUG0("ALSA loopback driver opened in Capture mode");

    /* Configure ALSA driver with PCM parameters */
    rv = snd_pcm_set_params(app_alsa_cb.p_capture_handle,
            format,
            access,
            nb_channels,
            p_open->sample_rate,
            1, /* SW resample */
            p_open->latency);
    if (rv)
    {
        APP_ERROR1("Unable to config ALSA device:%s", snd_strerror(rv));
        snd_pcm_close(app_alsa_cb.p_capture_handle);
        app_alsa_cb.p_capture_handle = NULL;
        return rv;
    }
    return 0;
}

int main(int argc, char **argv)
{
   register int  err;
   int           cardNum;

   // Start with first card
   cardNum = -1;

   for (;;)
   {
      snd_ctl_t *cardHandle;

      // Get next sound card's card number. When "cardNum" == -1, then ALSA
      // fetches the first card
      if ((err = snd_card_next(&cardNum)) < 0)
      {
         printf("Can't get the next card number: %s\n", snd_strerror(err));
         break;
      }

      // No more cards? ALSA sets "cardNum" to -1 if so
      if (cardNum < 0) break;

      // Open this card's control interface. We specify only the card number -- not
      // any device nor sub-device too
      {
      char   str[64];

      sprintf(str, "hw:%i", cardNum);
      if ((err = snd_ctl_open(&cardHandle, str, 0)) < 0)
      {
         printf("Can't open card %i: %s\n", cardNum, snd_strerror(err));
         continue;
      }
      }

      {
      int      devNum, totalDevices;

      // No waveform devices found yet
      totalDevices = 0;

      // Start with the first wave device on this card
      devNum = -1;
		
      for (;;)
      {
         // Get the number of the next wave device on this card
         if ((err = snd_ctl_pcm_next_device(cardHandle, &devNum)) < 0)
         {
            printf("Can't get next wave device number: %s\n", snd_strerror(err));
            break;
         }

         // No more wave devices on this card? ALSA sets "devNum" to -1 if so.
         // NOTE: It's possible that this sound card may have no wave devices on it
         // at all, for example if it's only a MIDI card
         if (devNum < 0) break;

         // Another wave device found on this card, so bump the count
         ++totalDevices;
      }

      printf("Found %i digital audio devices on card %i\n", totalDevices, cardNum);
      }

      // Close the card's control interface after we're done with it
      snd_ctl_close(cardHandle);
   }

   // ALSA allocates some mem to load its config file when we call some of the
   // above functions. Now that we're done getting the info, let's tell ALSA
   // to unload the info and free up that mem
   snd_config_update_free_global();
}

static int
mixer_open(void)
{
  snd_mixer_elem_t *elem;
  snd_mixer_elem_t *master;
  snd_mixer_elem_t *pcm;
  snd_mixer_elem_t *custom;
  snd_mixer_selem_id_t *sid;
  int ret;

  ret = snd_mixer_open(&mixer_hdl, 0);
  if (ret < 0)
    {
      DPRINTF(E_LOG, L_LAUDIO, "Failed to open mixer: %s\n", snd_strerror(ret));

      mixer_hdl = NULL;
      return -1;
    }

  ret = snd_mixer_attach(mixer_hdl, card_name);
  if (ret < 0)
    {
      DPRINTF(E_LOG, L_LAUDIO, "Failed to attach mixer: %s\n", snd_strerror(ret));

      goto out_close;
    }

  ret = snd_mixer_selem_register(mixer_hdl, NULL, NULL);
  if (ret < 0)
    {
      DPRINTF(E_LOG, L_LAUDIO, "Failed to register mixer: %s\n", snd_strerror(ret));

      goto out_detach;
    }

  ret = snd_mixer_load(mixer_hdl);
  if (ret < 0)
    {
      DPRINTF(E_LOG, L_LAUDIO, "Failed to load mixer: %s\n", snd_strerror(ret));

      goto out_detach;
    }

  /* Grab interesting elements */
  snd_mixer_selem_id_alloca(&sid);

  pcm = NULL;
  master = NULL;
  custom = NULL;
  for (elem = snd_mixer_first_elem(mixer_hdl); elem; elem = snd_mixer_elem_next(elem))
    {
      snd_mixer_selem_get_id(elem, sid);

      if (mixer_name && (strcmp(snd_mixer_selem_id_get_name(sid), mixer_name) == 0))
	{
	  custom = elem;
	  break;
	}
      else if (strcmp(snd_mixer_selem_id_get_name(sid), "PCM") == 0)
        pcm = elem;
      else if (strcmp(snd_mixer_selem_id_get_name(sid), "Master") == 0)
	master = elem;
    }

  if (mixer_name)
    {
      if (custom)
	vol_elem = custom;
      else
	{
	  DPRINTF(E_LOG, L_LAUDIO, "Failed to open configured mixer element '%s'\n", mixer_name);

	  goto out_detach;
	}
    }
  else if (pcm)
    vol_elem = pcm;
  else if (master)
    vol_elem = master;
  else
    {
      DPRINTF(E_LOG, L_LAUDIO, "Failed to open PCM or Master mixer element\n");

      goto out_detach;
    }

  /* Get min & max volume */
  snd_mixer_selem_get_playback_volume_range(vol_elem, &vol_min, &vol_max);

  return 0;

 out_detach:
  snd_mixer_detach(mixer_hdl, card_name);
 out_close:
  snd_mixer_close(mixer_hdl);
  mixer_hdl = NULL;
  vol_elem = NULL;

  return -1;
}

static int
laudio_alsa_open(void)
{
  snd_pcm_hw_params_t *hw_params;
  snd_pcm_uframes_t bufsize;
  snd_pcm_uframes_t period_size;
  int ret;

  hw_params = NULL;

  ret = snd_pcm_open(&hdl, card_name, SND_PCM_STREAM_PLAYBACK, 0);
  if (ret < 0)
    {
      DPRINTF(E_LOG, L_LAUDIO, "Could not open playback device: %s\n", snd_strerror(ret));

      return -1;
    }

  /* HW params */
  ret = snd_pcm_hw_params_malloc(&hw_params);
  if (ret < 0)
    {
      DPRINTF(E_LOG, L_LAUDIO, "Could not allocate hw params: %s\n", snd_strerror(ret));

      goto out_fail;
    }

  ret = snd_pcm_hw_params_any(hdl, hw_params);
  if (ret < 0)
    {
      DPRINTF(E_LOG, L_LAUDIO, "Could not retrieve hw params: %s\n", snd_strerror(ret));

      goto out_fail;
    }

  ret = snd_pcm_hw_params_set_access(hdl, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
  if (ret < 0)
    {
      DPRINTF(E_LOG, L_LAUDIO, "Could not set access method: %s\n", snd_strerror(ret));

      goto out_fail;
    }

  ret = snd_pcm_hw_params_set_format(hdl, hw_params, SND_PCM_FORMAT_S16_LE);
  if (ret < 0)
    {
      DPRINTF(E_LOG, L_LAUDIO, "Could not set S16LE format: %s\n", snd_strerror(ret));

      goto out_fail;
    }

  ret = snd_pcm_hw_params_set_channels(hdl, hw_params, 2);
  if (ret < 0)
    {
      DPRINTF(E_LOG, L_LAUDIO, "Could not set stereo output: %s\n", snd_strerror(ret));

      goto out_fail;
    }

  ret = snd_pcm_hw_params_set_rate(hdl, hw_params, 44100, 0);
  if (ret < 0)
    {
      DPRINTF(E_LOG, L_LAUDIO, "Hardware doesn't support 44.1 kHz: %s\n", snd_strerror(ret));

      goto out_fail;
    }

  ret = snd_pcm_hw_params_get_buffer_size_max(hw_params, &bufsize);
  if (ret < 0)
    {
      DPRINTF(E_LOG, L_LAUDIO, "Could not get max buffer size: %s\n", snd_strerror(ret));

      goto out_fail;
    }

  DPRINTF(E_DBG, L_LAUDIO, "Max buffer size is %lu samples\n", bufsize);

  ret = snd_pcm_hw_params_set_buffer_size_max(hdl, hw_params, &bufsize);
  if (ret < 0)
    {
      DPRINTF(E_LOG, L_LAUDIO, "Could not set buffer size to max: %s\n", snd_strerror(ret));

      goto out_fail;
    }

  // With a small period size we seem to get underruns because the period time
  // passes before we manage to feed with samples (if the player is slightly
  // behind - especially critical during startup when the buffer is low)
  // Internet suggests period_size should be /2 bufsize, but default seems to be
  // much lower, so compromise on /4 (but not more than 65536 frames = almost 2 sec).
  period_size = bufsize / 4;
  if (period_size > 65536)
    period_size = 65536;

  ret = snd_pcm_hw_params_set_period_size_near(hdl, hw_params, &period_size, NULL);
  if (ret < 0)
    {
      DPRINTF(E_LOG, L_LAUDIO, "Could not set period size: %s\n", snd_strerror(ret));

      goto out_fail;
//.........这里部分代码省略.........

static int
laudio_alsa_start(uint64_t cur_pos, uint64_t next_pkt)
{
  snd_output_t *output;
  char *debug_pcm_cfg;
  int ret;

  ret = snd_pcm_prepare(hdl);
  if (ret < 0)
    {
      DPRINTF(E_LOG, L_LAUDIO, "Could not prepare PCM device: %s\n", snd_strerror(ret));

      return -1;
    }

  DPRINTF(E_DBG, L_LAUDIO, "Start local audio curpos %" PRIu64 ", next_pkt %" PRIu64 "\n", cur_pos, next_pkt);

  /* Make pcm_pos the rtptime of the packet containing cur_pos */
  pcm_pos = next_pkt;
  while (pcm_pos > cur_pos)
    pcm_pos -= AIRTUNES_V2_PACKET_SAMPLES;

  pcm_start_pos = next_pkt + pcm_period_size;

  /* Compensate period size, otherwise get_pos won't be correct */
  pcm_pos += pcm_period_size;

  DPRINTF(E_DBG, L_LAUDIO, "PCM pos %" PRIu64 ", start pos %" PRIu64 "\n", pcm_pos, pcm_start_pos);

  pcm_pkt_head = NULL;
  pcm_pkt_tail = NULL;

  pcm_last_error = 0;
  pcm_recovery = 0;

  // alsa doesn't actually seem to wait for this threshold?
  ret = laudio_alsa_set_start_threshold(pcm_buf_threshold);
  if (ret < 0)
    {
      DPRINTF(E_LOG, L_LAUDIO, "Could not set PCM start threshold for local audio start\n");

      return -1;
    }

  // Dump PCM config data for E_DBG logging
  ret = snd_output_buffer_open(&output);
  if (ret == 0)
    {
      if (snd_pcm_dump_setup(hdl, output) == 0)
	{
	  snd_output_buffer_string(output, &debug_pcm_cfg);
	  DPRINTF(E_DBG, L_LAUDIO, "Dump of sound device config:\n%s\n", debug_pcm_cfg);
	}

      snd_output_close(output);
    }

  update_status(LAUDIO_STARTED);

  return 0;
}

static void
laudio_alsa_write(uint8_t *buf, uint64_t rtptime)
{
  struct pcm_packet *pkt;
  snd_pcm_sframes_t nsamp;
  int ret;

  pkt = (struct pcm_packet *)malloc(sizeof(struct pcm_packet));
  if (!pkt)
    {
      DPRINTF(E_LOG, L_LAUDIO, "Out of memory for PCM pkt\n");

      update_status(LAUDIO_FAILED);
      return;
    }

  memcpy(pkt->samples, buf, sizeof(pkt->samples));

  pkt->rtptime = rtptime;
  pkt->offset = 0;
  pkt->next = NULL;

  if (pcm_pkt_tail)
    {
      pcm_pkt_tail->next = pkt;
      pcm_pkt_tail = pkt;
    }
  else
    {
      pcm_pkt_head = pkt;
      pcm_pkt_tail = pkt;
    }

  if (pcm_pos < pcm_pkt_head->rtptime)
    {
      pcm_pos += AIRTUNES_V2_PACKET_SAMPLES;

      return;
    }
  else if ((pcm_status != LAUDIO_RUNNING) && (pcm_pos >= pcm_start_pos))
    {
      update_status(LAUDIO_RUNNING);
    }

  pkt = pcm_pkt_head;

  while (pkt)
    {
      if (pcm_recovery)
	{
	  ret = laudio_alsa_xrun_recover(0);
	  if ((ret == 2) && (pcm_recovery < 10))
	    return;
	  else
	    {
	      if (ret == 2)
		DPRINTF(E_LOG, L_LAUDIO, "Couldn't recover PCM device after 10 tries, aborting\n");

	      update_status(LAUDIO_FAILED);
	      return;
	    }
	}

      nsamp = snd_pcm_writei(hdl, pkt->samples + pkt->offset, BTOS(sizeof(pkt->samples) - pkt->offset));
      if ((nsamp == -EPIPE) || (nsamp == -ESTRPIPE))
	{
	  ret = laudio_alsa_xrun_recover(nsamp);
	  if ((ret < 0) || (ret == 1))
	    {
	      if (ret < 0)
		DPRINTF(E_LOG, L_LAUDIO, "PCM write error: %s\n", snd_strerror(ret));

	      update_status(LAUDIO_FAILED);
	      return;
	    }
	  else if (ret != 0)
	    return;

	  continue;
	}
      else if (nsamp < 0)
	{
	  DPRINTF(E_LOG, L_LAUDIO, "PCM write error: %s\n", snd_strerror(nsamp));

	  update_status(LAUDIO_FAILED);
	  return;
	}

      pcm_last_error = 0;
      pcm_pos += nsamp;

      pkt->offset += STOB(nsamp);
      if (pkt->offset == sizeof(pkt->samples))
	{
	  pcm_pkt_head = pkt->next;

	  if (pkt == pcm_pkt_tail)
	    pcm_pkt_tail = NULL;

	  free(pkt);
//.........这里部分代码省略.........

/**************************************************************************
 * 				wodOpen				[internal]
 */
static DWORD wodOpen(WORD wDevID, LPWAVEOPENDESC lpDesc, DWORD dwFlags)
{
    WINE_WAVEDEV*	        wwo;
    snd_pcm_t *                 pcm = NULL;
    snd_hctl_t *                hctl = NULL;
    snd_pcm_hw_params_t *       hw_params = NULL;
    snd_pcm_sw_params_t *       sw_params;
    snd_pcm_access_t            access;
    snd_pcm_format_t            format = -1;
    unsigned int                rate;
    unsigned int                buffer_time = 120000;
    unsigned int                period_time = 22000;
    snd_pcm_uframes_t           buffer_size;
    snd_pcm_uframes_t           period_size;
    int                         flags;
    int                         err=0;
    int                         dir=0;
    DWORD                       retcode = 0;

    TRACE("(%u, %p, %08X);\n", wDevID, lpDesc, dwFlags);
    if (lpDesc == NULL) {
	WARN("Invalid Parameter !\n");
	return MMSYSERR_INVALPARAM;
    }
    if (wDevID >= ALSA_WodNumDevs) {
	TRACE("Asked for device %d, but only %d known!\n", wDevID, ALSA_WodNumDevs);
	return MMSYSERR_BADDEVICEID;
    }

    /* only PCM format is supported so far... */
    if (!ALSA_supportedFormat(lpDesc->lpFormat)) {
	WARN("Bad format: tag=%04X nChannels=%d nSamplesPerSec=%d !\n",
	     lpDesc->lpFormat->wFormatTag, lpDesc->lpFormat->nChannels,
	     lpDesc->lpFormat->nSamplesPerSec);
	return WAVERR_BADFORMAT;
    }

    if (dwFlags & WAVE_FORMAT_QUERY) {
	TRACE("Query format: tag=%04X nChannels=%d nSamplesPerSec=%d !\n",
	     lpDesc->lpFormat->wFormatTag, lpDesc->lpFormat->nChannels,
	     lpDesc->lpFormat->nSamplesPerSec);
	return MMSYSERR_NOERROR;
    }

    wwo = &WOutDev[wDevID];

    if (wwo->pcm != NULL) {
        WARN("%d already allocated\n", wDevID);
        return MMSYSERR_ALLOCATED;
    }

    if (dwFlags & WAVE_DIRECTSOUND)
        FIXME("Why are we called with DirectSound flag? It doesn't use MMSYSTEM any more\n");
        /* not supported, ignore it */
    dwFlags &= ~WAVE_DIRECTSOUND;

    flags = SND_PCM_NONBLOCK;

    if ( (err = snd_pcm_open(&pcm, wwo->pcmname, SND_PCM_STREAM_PLAYBACK, flags)) < 0)
    {
        ERR("Error open: %s\n", snd_strerror(err));
	return MMSYSERR_NOTENABLED;
    }

    if (wwo->ctlname)
    {
        err = snd_hctl_open(&hctl, wwo->ctlname, 0);
        if (err >= 0)
        {
            snd_hctl_load(hctl);
        }
        else
        {
            WARN("Could not open hctl for [%s]: %s\n", wwo->ctlname, snd_strerror(err));
            hctl = NULL;
        }
    }

    wwo->wFlags = HIWORD(dwFlags & CALLBACK_TYPEMASK);

    wwo->waveDesc = *lpDesc;
    ALSA_copyFormat(lpDesc->lpFormat, &wwo->format);

    TRACE("Requested this format: %dx%dx%d %s\n",
          wwo->format.Format.nSamplesPerSec,
          wwo->format.Format.wBitsPerSample,
          wwo->format.Format.nChannels,
          ALSA_getFormat(wwo->format.Format.wFormatTag));

    if (wwo->format.Format.wBitsPerSample == 0) {
	WARN("Resetting zeroed wBitsPerSample\n");
	wwo->format.Format.wBitsPerSample = 8 *
	    (wwo->format.Format.nAvgBytesPerSec /
	     wwo->format.Format.nSamplesPerSec) /
	    wwo->format.Format.nChannels;
    }

//.........这里部分代码省略.........

//.........这里部分代码省略.........
	  break;
	case 2:
	  alsa_fragcount = 8;
	  chunk_size = 1024;
	    mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 8192\n");
	    mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 1024\n");
	  break;
	case 3:
	  alsa_fragcount = 32;
	  chunk_size = 512;
	    mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 16384\n");
	    mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 512\n");
	  break;
	case 4:
	  alsa_fragcount = 16;
	  chunk_size = 1024;
	    mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 16384\n");
	    mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 1024\n");
	  break;
	default:
	  alsa_fragcount = 16;
	  chunk_size = 1024;
	  break;
	}
    }

    if (!alsa_handler) {
      //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
      if ((err = try_open_device(alsa_device, open_mode, format == AF_FORMAT_AC3)) < 0)
	{
	  if (err != -EBUSY && ao_noblock) {
	    mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_OpenInNonblockModeFailed);
	    if ((err = try_open_device(alsa_device, 0, format == AF_FORMAT_AC3)) < 0) {
	      mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err));
	      return(0);
	    }
	  } else {
	    mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err));
	    return(0);
	  }
	}

      if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) {
         mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSetBlockMode, snd_strerror(err));
      } else {
	mp_msg(MSGT_AO,MSGL_V,"alsa-init: pcm opened in blocking mode\n");
      }

      snd_pcm_hw_params_alloca(&alsa_hwparams);
      snd_pcm_sw_params_alloca(&alsa_swparams);

      // setting hw-parameters
      if ((err = snd_pcm_hw_params_any(alsa_handler, alsa_hwparams)) < 0)
	{
	  mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetInitialParameters,
		 snd_strerror(err));
	  return(0);
	}
    
      err = snd_pcm_hw_params_set_access(alsa_handler, alsa_hwparams,
					 SND_PCM_ACCESS_RW_INTERLEAVED);
      if (err < 0) {
	mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetAccessType, 
	       snd_strerror(err));
	return (0);
      }

static void *alsa_qsa_init(const char *device,
      unsigned rate, unsigned latency)
{
   int err, card, dev, i;
   snd_pcm_channel_params_t params = {0};
   snd_pcm_channel_info_t pi;
   snd_pcm_channel_setup_t setup = {0};
   settings_t *settings = config_get_ptr();
   alsa_t *alsa = (alsa_t*)calloc(1, sizeof(alsa_t));
   if (!alsa)
      return NULL;

   (void)device;
   (void)rate;
   (void)latency;

   if ((err = snd_pcm_open_preferred(&alsa->pcm, &card, &dev,
               SND_PCM_OPEN_PLAYBACK)) < 0)
   {
      RARCH_ERR("[ALSA QSA]: Audio open error: %s\n",
            snd_strerror(err));
      goto error;
   }

   if((err = snd_pcm_nonblock_mode(alsa->pcm, 1)) < 0)
   {
      RARCH_ERR("[ALSA QSA]: Can't set blocking mode: %s\n",
            snd_strerror(err));
      goto error;
   }

   memset(&pi, 0, sizeof(pi));
   pi.channel = SND_PCM_CHANNEL_PLAYBACK;
   if ((err = snd_pcm_channel_info(alsa->pcm, &pi)) < 0)
   {
      RARCH_ERR("[ALSA QSA]: snd_pcm_channel_info failed: %s\n",
            snd_strerror(err));
      goto error;
   }

   memset(&params, 0, sizeof(params));

   params.channel = SND_PCM_CHANNEL_PLAYBACK;
   params.mode = SND_PCM_MODE_BLOCK;

   params.format.interleave = 1;
   params.format.format = SND_PCM_SFMT_S16_LE;
   params.format.rate = DEFAULT_RATE;
   params.format.voices = 2;

   params.start_mode = SND_PCM_START_FULL;
   params.stop_mode = SND_PCM_STOP_STOP;

   params.buf.block.frag_size = pi.max_fragment_size;
   params.buf.block.frags_min = 2;
   params.buf.block.frags_max = 8;

   RARCH_LOG("Fragment size: %d\n", params.buf.block.frag_size);
   RARCH_LOG("Min Fragment size: %d\n", params.buf.block.frags_min);
   RARCH_LOG("Max Fragment size: %d\n", params.buf.block.frags_max);

   if ((err = snd_pcm_channel_params(alsa->pcm, &params)) < 0)
   {
      RARCH_ERR("[ALSA QSA]: Channel Parameter Error: %s\n",
            snd_strerror(err));
      goto error;
   }

   setup.channel = SND_PCM_CHANNEL_PLAYBACK;

   if ((err = snd_pcm_channel_setup(alsa->pcm, &setup)) < 0)
   {
      RARCH_ERR("[ALSA QSA]: Channel Parameter Read Back Error: %s\n",
            snd_strerror(err));
      goto error;
   }

   if (settings->audio.block_frames)
      alsa->buf_size = settings->audio.block_frames * 4;
   else
      alsa->buf_size = next_pow2(32 * latency);

   RARCH_LOG("[ALSA QSA]: buffer size: %u bytes\n", alsa->buf_size);

   alsa->buf_count = (latency * 4 * rate + 500) / 1000;
   alsa->buf_count = (alsa->buf_count + alsa->buf_size / 2) / alsa->buf_size;

   if ((err = snd_pcm_channel_prepare(alsa->pcm,
               SND_PCM_CHANNEL_PLAYBACK)) < 0)
   {
      RARCH_ERR("[ALSA QSA]: Channel Prepare Error: %s\n",
            snd_strerror(err));
      goto error;
   }

   alsa->buffer = (uint8_t**)calloc(sizeof(uint8_t*), alsa->buf_count);
   if (!alsa->buffer)
      goto error;

   alsa->buffer_chunk = (uint8_t*)calloc(alsa->buf_count, alsa->buf_size);
//.........这里部分代码省略.........