main() {
  long loops;
  int rc;
  int size;
  snd_pcm_t *handle;
  snd_pcm_hw_params_t *params;
  unsigned int val;
  int dir;
  snd_pcm_uframes_t frames;
 unsigned char *buffer;
  int i=0;

/* socket setting */
   int sd;
   struct sockaddr_in s_addr;
   int n, n_send, status;

   int on = 1;


   sd = socket (AF_INET, SOCK_DGRAM, 0);

   bzero(&s_addr, sizeof(s_addr));
   s_addr.sin_family = AF_INET;
   s_addr.sin_addr.s_addr = inet_addr("192.168.42.255");
   s_addr.sin_port = htons(2007);

   if((status = setsockopt(sd, SOL_SOCKET, SO_BROADCAST, &on, sizeof(on))) != 0 )
   {
      printf("setsockopt error\n");
      exit(-1);
   }


  /* Open PCM device for recording (capture). */
  rc = snd_pcm_open(&handle, "plughw:1,0",
                    SND_PCM_STREAM_CAPTURE, 0);
  if (rc < 0) {
    fprintf(stderr,
            "unable to open pcm device: %s\n",
            snd_strerror(rc));
    exit(1);
  }

  /* Allocate a hardware parameters object. */
  snd_pcm_hw_params_alloca(&params);

  /* Fill it in with default values. */
  snd_pcm_hw_params_any(handle, params);

  /* Set the desired hardware parameters. */

  /* Interleaved mode */
  snd_pcm_hw_params_set_access(handle, params,
                      SND_PCM_ACCESS_RW_INTERLEAVED);

  /* Signed 16-bit little-endian format */
  snd_pcm_hw_params_set_format(handle, params,SND_PCM_FORMAT_U8);

  /* Two channels (stereo) */
  snd_pcm_hw_params_set_channels(handle, params, 1);

  /* 44100 bits/second sampling rate (CD quality) */
  val = 44100;
  snd_pcm_hw_params_set_rate_near(handle, params,
                                  &val, &dir);

  /* Set period size to 32 frames. */
  frames =4800;
  snd_pcm_hw_params_set_period_size_near(handle,
                              params, &frames, &dir);

  /* Write the parameters to the driver */
  rc = snd_pcm_hw_params(handle, params);
  if (rc < 0) {
    fprintf(stderr,
            "unable to set hw parameters: %s\n",
            snd_strerror(rc));
    exit(1);
  }

  /* Use a buffer large enough to hold one period */
  snd_pcm_hw_params_get_period_size(params,
                                      &frames, &dir);
  size = frames *1; /* 2 bytes/sample, 2 channels */

  buffer = (unsigned char*) malloc(size);
	
  /* We want to loop for 5 seconds */
  snd_pcm_hw_params_get_period_time(params,
                                         &val, &dir);
  while (1) {
    rc = snd_pcm_readi(handle, buffer, frames);
    if (rc == -EPIPE) {
      /* EPIPE means overrun */
      fprintf(stderr, "overrun occurred\n");
      snd_pcm_prepare(handle);
    } else if (rc < 0) {
      fprintf(stderr,
              "error from read: %s\n",
//.........这里部分代码省略.........

bool OutputALSA::initialize(quint32 freq, ChannelMap map, Qmmp::AudioFormat format)
{
    m_inited = false;

    if (pcm_handle)
        return false;

    if (snd_pcm_open(&pcm_handle, pcm_name, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK) < 0)
    {
        qWarning ("OutputALSA: Error opening PCM device %s", pcm_name);
        return false;
    }

    // we need to configure

    uint rate = freq; /* Sample rate */
    uint exact_rate = freq;   /* Sample rate returned by */

    /* load settings from config */
    QSettings settings(Qmmp::configFile(), QSettings::IniFormat);
    settings.beginGroup("ALSA");
    uint buffer_time = settings.value("buffer_time",500).toUInt()*1000;
    uint period_time = settings.value("period_time",100).toUInt()*1000;
    bool use_pause =  settings.value("use_snd_pcm_pause", false).toBool();
    settings.endGroup();

    snd_pcm_hw_params_t *hwparams = 0;
    snd_pcm_sw_params_t *swparams = 0;
    int err; //alsa error code

    //hw params
    snd_pcm_hw_params_alloca(&hwparams);
    if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0)
    {
        qWarning("OutputALSA: Can not read configuration for PCM device: %s", snd_strerror(err));
        return false;
    }
    if (m_use_mmap)
    {
        if ((err = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_MMAP_INTERLEAVED)) < 0)
        {
            qWarning("OutputALSA: Error setting mmap access: %s", snd_strerror(err));
            m_use_mmap = false;
        }
    }
    if (!m_use_mmap)
    {
        if ((err = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
        {
            qWarning("OutputALSA: Error setting access: %s", snd_strerror(err));
            return false;
        }
    }
    snd_pcm_format_t alsa_format = SND_PCM_FORMAT_UNKNOWN;
    switch (format)
    {
    case Qmmp::PCM_S8:
        alsa_format = SND_PCM_FORMAT_S8;
        break;
    case Qmmp::PCM_S16LE:
        alsa_format = SND_PCM_FORMAT_S16_LE;
        break;
    case Qmmp::PCM_S24LE:
        alsa_format = SND_PCM_FORMAT_S24_LE;
        break;
    case Qmmp::PCM_S32LE:
        alsa_format = SND_PCM_FORMAT_S32_LE;
        break;
    default:
        qWarning("OutputALSA: unsupported format detected");
        return false;
    }
    if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, alsa_format)) < 0)
    {
        qDebug("OutputALSA: Error setting format: %s", snd_strerror(err));
        return false;
    }
    exact_rate = rate;

    if ((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &exact_rate, 0)) < 0)
    {
        qWarning("OutputALSA: Error setting rate: %s", snd_strerror(err));
        return false;
    }
    if (rate != exact_rate)
    {
        qWarning("OutputALSA: The rate %d Hz is not supported by your hardware.\n==> Using %d Hz instead.", rate, exact_rate);
        rate = exact_rate;
    }
    uint c = map.count();
    if ((err = snd_pcm_hw_params_set_channels_near(pcm_handle, hwparams, &c)) < 0)
    {
        qWarning("OutputALSA: Error setting channels: %s", snd_strerror(err));
        return false;
    }
    if (c != (uint)map.count())
    {
        qWarning("OutputALSA: The channel number %d is not supported by your hardware", map.count());
        qWarning("==> Using %d instead.", c);
    }
//.........这里部分代码省略.........

//.........这里部分代码省略.........
      if (!info.m_displayNameExtra.empty())
        info.m_displayNameExtra += ' ';
      info.m_displayNameExtra += "S/PDIF";
    }
    else if (info.m_displayNameExtra.empty())
    {
      /* for USB audio, it gets a bit confusing as there is
       * - "SB Live! 24-bit External"
       * - "SB Live! 24-bit External, S/PDIF"
       * so add "Analog" qualifier to the first one */
      info.m_displayNameExtra = "Analog";
    }

    /* "default" is a device that will be used for all inputs, while
     * "@" will be mangled to front/default/surroundXX as necessary */
    if (device == "@" || device == "default")
    {
      /* Make it "Default (whatever)" */
      info.m_displayName = "Default (" + info.m_displayName + (info.m_displayNameExtra.empty() ? "" : " " + info.m_displayNameExtra + ")");
      info.m_displayNameExtra = "";
    }

  }
  else
  {
    /* virtual devices: "default", "pulse", ... */
    /* description can be e.g. "PulseAudio Sound Server" - for hw devices it is
     * normally uninteresting, like "HDMI Audio Output" or "Default Audio Device",
     * so we only use it for virtual devices that have no better display name */
    info.m_displayName = description;
  }

  snd_pcm_hw_params_t *hwparams;
  snd_pcm_hw_params_alloca(&hwparams);
  memset(hwparams, 0, snd_pcm_hw_params_sizeof());

  /* ensure we can get a playback configuration for the device */
  if (snd_pcm_hw_params_any(pcmhandle, hwparams) < 0)
  {
    CLog::Log(LOGINFO, "CAESinkALSA - No playback configurations available for device \"%s\"", device.c_str());
    snd_pcm_close(pcmhandle);
    return;
  }

  /* detect the available sample rates */
  for (unsigned int *rate = ALSASampleRateList; *rate != 0; ++rate)
    if (snd_pcm_hw_params_test_rate(pcmhandle, hwparams, *rate, 0) >= 0)
      info.m_sampleRates.push_back(*rate);

  /* detect the channels available */
  int channels = 0;
  for (int i = ALSA_MAX_CHANNELS; i >= 1; --i)
  {
    /* Reopen the device if needed on the special "surroundXX" cases */
    if (info.m_deviceType == AE_DEVTYPE_PCM && (i == 8 || i == 6 || i == 4))
      OpenPCMDevice(device, "", i, &pcmhandle, config, false);

    if (snd_pcm_hw_params_test_channels(pcmhandle, hwparams, i) >= 0)
    {
      channels = i;
      break;
    }
  }

  if (device == "default" && channels == 2)
  {
    /* This looks like the ALSA standard default stereo dmix device, we
     * probably want to use "@" instead to get surroundXX. */
    snd_pcm_close(pcmhandle);
    EnumerateDevice(list, "@", description, config);
    return;
  }

  CAEChannelInfo alsaChannels;
  for (int i = 0; i < channels; ++i)
  {
    if (!info.m_channels.HasChannel(ALSAChannelMap[i]))
      info.m_channels += ALSAChannelMap[i];
    alsaChannels += ALSAChannelMap[i];
  }

  /* remove the channels from m_channels that we cant use */
  info.m_channels.ResolveChannels(alsaChannels);

  /* detect the PCM sample formats that are available */
  for (enum AEDataFormat i = AE_FMT_MAX; i > AE_FMT_INVALID; i = (enum AEDataFormat)((int)i - 1))
  {
    if (AE_IS_RAW(i) || i == AE_FMT_MAX)
      continue;
    snd_pcm_format_t fmt = AEFormatToALSAFormat(i);
    if (fmt == SND_PCM_FORMAT_UNKNOWN)
      continue;

    if (snd_pcm_hw_params_test_format(pcmhandle, hwparams, fmt) >= 0)
      info.m_dataFormats.push_back(i);
  }

  snd_pcm_close(pcmhandle);
  list.push_back(info);
}

static snd_pcm_t *
alsa_open (int channels, unsigned samplerate, int realtime)
{	const char * device = "default" ;
	snd_pcm_t *alsa_dev = NULL ;
	snd_pcm_hw_params_t *hw_params ;
	snd_pcm_uframes_t buffer_size ;
	snd_pcm_uframes_t alsa_period_size, alsa_buffer_frames ;
	snd_pcm_sw_params_t *sw_params ;

	int err ;

	if (realtime)
	{	alsa_period_size = 256 ;
		alsa_buffer_frames = 3 * alsa_period_size ;
		}
	else
	{	alsa_period_size = 1024 ;
		alsa_buffer_frames = 4 * alsa_period_size ;
		} ;

	if ((err = snd_pcm_open (&alsa_dev, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0)
	{	fprintf (stderr, "cannot open audio device \"%s\" (%s)\n", device, snd_strerror (err)) ;
		goto catch_error ;
		} ;

	snd_pcm_nonblock (alsa_dev, 0) ;

	if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0)
	{	fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n", snd_strerror (err)) ;
		goto catch_error ;
		} ;

	if ((err = snd_pcm_hw_params_any (alsa_dev, hw_params)) < 0)
	{	fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n", snd_strerror (err)) ;
		goto catch_error ;
		} ;

	if ((err = snd_pcm_hw_params_set_access (alsa_dev, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
	{	fprintf (stderr, "cannot set access type (%s)\n", snd_strerror (err)) ;
		goto catch_error ;
		} ;

	if ((err = snd_pcm_hw_params_set_format (alsa_dev, hw_params, SND_PCM_FORMAT_FLOAT)) < 0)
	{	fprintf (stderr, "cannot set sample format (%s)\n", snd_strerror (err)) ;
		goto catch_error ;
		} ;

	if ((err = snd_pcm_hw_params_set_rate_near (alsa_dev, hw_params, &samplerate, 0)) < 0)
	{	fprintf (stderr, "cannot set sample rate (%s)\n", snd_strerror (err)) ;
		goto catch_error ;
		} ;

	if ((err = snd_pcm_hw_params_set_channels (alsa_dev, hw_params, channels)) < 0)
	{	fprintf (stderr, "cannot set channel count (%s)\n", snd_strerror (err)) ;
		goto catch_error ;
		} ;

	if ((err = snd_pcm_hw_params_set_buffer_size_near (alsa_dev, hw_params, &alsa_buffer_frames)) < 0)
	{	fprintf (stderr, "cannot set buffer size (%s)\n", snd_strerror (err)) ;
		goto catch_error ;
		} ;

	if ((err = snd_pcm_hw_params_set_period_size_near (alsa_dev, hw_params, &alsa_period_size, 0)) < 0)
	{	fprintf (stderr, "cannot set period size (%s)\n", snd_strerror (err)) ;
		goto catch_error ;
		} ;

	if ((err = snd_pcm_hw_params (alsa_dev, hw_params)) < 0)
	{	fprintf (stderr, "cannot set parameters (%s)\n", snd_strerror (err)) ;
		goto catch_error ;
		} ;

	/* extra check: if we have only one period, this code won't work */
	snd_pcm_hw_params_get_period_size (hw_params, &alsa_period_size, 0) ;
	snd_pcm_hw_params_get_buffer_size (hw_params, &buffer_size) ;
	if (alsa_period_size == buffer_size)
	{	fprintf (stderr, "Can't use period equal to buffer size (%lu == %lu)", alsa_period_size, buffer_size) ;
		goto catch_error ;
		} ;

	snd_pcm_hw_params_free (hw_params) ;

	if ((err = snd_pcm_sw_params_malloc (&sw_params)) != 0)
	{	fprintf (stderr, "%s: snd_pcm_sw_params_malloc: %s", __func__, snd_strerror (err)) ;
		goto catch_error ;
		} ;

	if ((err = snd_pcm_sw_params_current (alsa_dev, sw_params)) != 0)
	{	fprintf (stderr, "%s: snd_pcm_sw_params_current: %s", __func__, snd_strerror (err)) ;
		goto catch_error ;
		} ;

	/* note: set start threshold to delay start until the ring buffer is full */
	snd_pcm_sw_params_current (alsa_dev, sw_params) ;

	if ((err = snd_pcm_sw_params_set_start_threshold (alsa_dev, sw_params, buffer_size)) < 0)
	{	fprintf (stderr, "cannot set start threshold (%s)\n", snd_strerror (err)) ;
		goto catch_error ;
		} ;

//.........这里部分代码省略.........

int initAlsa(char **argv,int optind)
{
    snd_pcm_hw_params_t *hw_params;
    int err,n;
    unsigned int Fs;

    if ((err = snd_pcm_open(&capture_handle, argv[optind],
                            SND_PCM_STREAM_CAPTURE, 0)) < 0) {
        fprintf(stderr, "Alsa cannot open audio device %s (%s)\n",argv[optind], snd_strerror(err));
        return 1;
    }

    if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0) {
        fprintf(stderr,
                "Alsa cannot allocate hardware parameter structure (%s)\n",snd_strerror(err));
        return 1;
    }

    if ((err = snd_pcm_hw_params_any(capture_handle, hw_params)) < 0) {
        fprintf(stderr,
                "Alsa cannot initialize hardware parameter structure (%s)\n",snd_strerror(err));
        return 1;
    }

    if ((err = snd_pcm_hw_params_set_access(capture_handle, hw_params,SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
        fprintf(stderr, "Alsa cannot set access type (%s)\n",snd_strerror(err));
        return 1;
    }

    if ((err = snd_pcm_hw_params_set_format(capture_handle, hw_params,SND_PCM_FORMAT_S16)) < 0) {
        fprintf(stderr, "Alsa cannot set sample format (%s)\n",snd_strerror(err));
        return 1;
    }

    snd_pcm_hw_params_set_rate_resample(capture_handle, hw_params,0);
    Fs=19200;
    n=1;
    if ((err = snd_pcm_hw_params_set_rate_near(capture_handle, hw_params, &Fs,&n)) < 0) {
        fprintf(stderr, "Alsa cannot set sample rate (%s)\n",snd_strerror(err));
        return 1;
    }
    fprintf(stderr, "Alsa sample rate %d\n",Fs);

    if(snd_pcm_hw_params_get_channels (hw_params, &nbch)!=0) {
        fprintf(stderr, "Alsa cannot get number of channels\n");
        return 1;
    }
    if(nbch>4) {
        fprintf(stderr, "Alsa too much channels\n");
        return 1;

    }
    if ((err = snd_pcm_hw_params(capture_handle, hw_params)) < 0) {
        fprintf(stderr, "Alsa cannot set parameters (%s)\n",snd_strerror(err));
        return 1;
    }
    snd_pcm_hw_params_free(hw_params);

    if ((err = snd_pcm_prepare(capture_handle)) < 0) {
        fprintf(stderr,
                "Alsa cannot prepare audio interface for use (%s)\n",snd_strerror(err));
        return 1;
    }

    for(n=0; n<nbch; n++) {
        channel[n].chn=n;
        channel[n].Infs=Fs;
        channel[n].InBuff=malloc(MAXNBFRAMES*sizeof(sample_t));
    }
    for(; n<MAXNBCHANNELS; n++) channel[n].Infs=0;

    return (0);
}

    /**
     * Create and initialize Alsa audio output device with given parameters.
     *
     * @param Parameters - optional parameters
     * @throws AudioOutputException  if output device cannot be opened
     */
    AudioOutputDeviceAlsa::AudioOutputDeviceAlsa(std::map<String,DeviceCreationParameter*> Parameters) : AudioOutputDevice(Parameters), Thread(true, true, 1, 0) {
        pcm_handle           = NULL;
        stream               = SND_PCM_STREAM_PLAYBACK;
        this->uiAlsaChannels = ((DeviceCreationParameterInt*)Parameters["CHANNELS"])->ValueAsInt();
        this->uiSamplerate   = ((DeviceCreationParameterInt*)Parameters["SAMPLERATE"])->ValueAsInt();
        this->FragmentSize   = ((DeviceCreationParameterInt*)Parameters["FRAGMENTSIZE"])->ValueAsInt();
        uint Fragments       = ((DeviceCreationParameterInt*)Parameters["FRAGMENTS"])->ValueAsInt();
        String Card          = ((DeviceCreationParameterString*)Parameters["CARD"])->ValueAsString();

        dmsg(2,("Checking if hw parameters supported...\n"));
        if (HardwareParametersSupported(Card, uiAlsaChannels, uiSamplerate, Fragments, FragmentSize)) {
            pcm_name = "hw:" + Card;
        }
        else {
            fprintf(stderr, "Warning: your soundcard doesn't support chosen hardware parameters; ");
            fprintf(stderr, "trying to compensate support lack with plughw...");
            fflush(stdout);
            pcm_name = "plughw:" + Card;
        }
        dmsg(2,("HW check completed.\n"));

        int err;

        snd_pcm_hw_params_alloca(&hwparams);  // Allocate the snd_pcm_hw_params_t structure on the stack.

        /* Open PCM. The last parameter of this function is the mode. */
        /* If this is set to 0, the standard mode is used. Possible   */
        /* other values are SND_PCM_NONBLOCK and SND_PCM_ASYNC.       */
        /* If SND_PCM_NONBLOCK is used, read / write access to the    */
        /* PCM device will return immediately. If SND_PCM_ASYNC is    */
        /* specified, SIGIO will be emitted whenever a period has     */
        /* been completely processed by the soundcard.                */
        if ((err = snd_pcm_open(&pcm_handle, pcm_name.c_str(), stream, 0)) < 0) {
            throw AudioOutputException(String("Error opening PCM device ") + pcm_name + ": " + snd_strerror(err));
        }

        if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0) {
            throw AudioOutputException(String("Error, cannot initialize hardware parameter structure: ") + snd_strerror(err));
        }

        /* Set access type. This can be either    */
        /* SND_PCM_ACCESS_RW_INTERLEAVED or       */
        /* SND_PCM_ACCESS_RW_NONINTERLEAVED.      */
        if ((err = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
            throw AudioOutputException(String("Error snd_pcm_hw_params_set_access: ") + snd_strerror(err));
        }

        /* Set sample format */
        #if WORDS_BIGENDIAN
        if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_BE)) < 0)
        #else // little endian
        if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE)) < 0)
        #endif
        {
            throw AudioOutputException(String("Error setting sample format: ") + snd_strerror(err));
        }

        int dir = 0;

        /* Set sample rate. If the exact rate is not supported */
        /* by the hardware, use nearest possible rate.         */
        #if ALSA_MAJOR > 0
        if((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &uiSamplerate, &dir)) < 0)
        #else
        if((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, uiSamplerate, &dir)) < 0)
        #endif
        {
            throw AudioOutputException(String("Error setting sample rate: ") + snd_strerror(err));
        }

        if ((err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, uiAlsaChannels)) < 0) {
            throw AudioOutputException(String("Error setting number of channels: ") + snd_strerror(err));
        }

        /* Set number of periods. Periods used to be called fragments. */
        if ((err = snd_pcm_hw_params_set_periods(pcm_handle, hwparams, Fragments, dir)) < 0) {
            throw AudioOutputException(String("Error setting number of ") + ToString(Fragments) + " periods: " + snd_strerror(err));
        }

        /* Set buffer size (in frames). The resulting latency is given by */
        /* latency = periodsize * periods / (rate * bytes_per_frame)     */
        if ((err = snd_pcm_hw_params_set_buffer_size(pcm_handle, hwparams, (FragmentSize * Fragments))) < 0) {
            throw AudioOutputException(String("Error setting buffersize: ") + snd_strerror(err));
        }

        /* Apply HW parameter settings to */
        /* PCM device and prepare device  */
        if ((err = snd_pcm_hw_params(pcm_handle, hwparams)) < 0) {
            throw AudioOutputException(String("Error setting HW params: ") + snd_strerror(err));
        }

        if (snd_pcm_sw_params_malloc(&swparams) != 0) {
            throw AudioOutputException(String("Error in snd_pcm_sw_params_malloc: ") + snd_strerror(err));
        }
//.........这里部分代码省略.........

/* Open and init the default recording device. */
void audioInit(void)
{
	int rc;
	int size;
	snd_pcm_hw_params_t *params;
	unsigned int val;
	int dir = 0;

	/* Open PCM device for recording (capture). */
	rc = snd_pcm_open(&sound.handle, getenvDefault(SOUND_DEVICE_ENV,
	                                               SOUND_DEVICE),
	                  SND_PCM_STREAM_CAPTURE, 0);
	if (rc < 0)
	{
		fprintf(stderr, "unable to open pcm device: %s\n", snd_strerror(rc));
		exit(EXIT_FAILURE);
	}

	/* Allocate a hardware parameters object. */
	snd_pcm_hw_params_alloca(&params);

	/* Fill it in with default values. */
	snd_pcm_hw_params_any(sound.handle, params);

	/* Set the desired hardware parameters. */

	/* Interleaved mode. */
	snd_pcm_hw_params_set_access(sound.handle, params,
	                             SND_PCM_ACCESS_RW_INTERLEAVED);

	/* Signed 16-bit little-endian format. */
	snd_pcm_hw_params_set_format(sound.handle, params,
	                             SND_PCM_FORMAT_S16_LE);

	/* One channel (mono). */
	snd_pcm_hw_params_set_channels(sound.handle, params, 1);

	/* 44100 bits/second sampling rate (CD quality). */
	val = SOUND_RATE;
	snd_pcm_hw_params_set_rate_near(sound.handle, params, &val, &dir);

	/* Set period size. It's best to set this to the same value as
	 * SOUND_SAMPLES_PER_TURN. A lower value would generate more
	 * hardware interrupts and thus a lower latency but that's of no use
	 * since we have to wait for an amount of SOUND_SAMPLES_PER_TURN
	 * samples anyway. */
	snd_pcm_uframes_t frames = SOUND_SAMPLES_PER_TURN;
	snd_pcm_hw_params_set_period_size_near(sound.handle, params,
	                                       &frames, &dir);

	/* Write the parameters to the driver. */
	rc = snd_pcm_hw_params(sound.handle, params);
	if (rc < 0)
	{
		fprintf(stderr, "unable to set hw parameters: %s\n",
		        snd_strerror(rc));
		exit(EXIT_FAILURE);
	}

	/* Acquire n frames per turn. */
	sound.bufferSizeFrames = SOUND_SAMPLES_PER_TURN;
	size = sound.bufferSizeFrames * 2;  /* 2 bytes/sample, 1 channel */

	/* Initialize the buffer. */
	sound.buffer = (char *)malloc(size);
	sound.bufferLast = (char *)malloc(size);
	sound.bufferFill = 0;
	sound.bufferReady = 0;

	/* Try to switch to non-blocking mode for reading. If that fails,
	 * print a warning and continue in blocking mode. */
	rc = snd_pcm_nonblock(sound.handle, 1);
	if (rc < 0)
	{
		fprintf(stderr, "Could not switch to non-blocking mode: %s\n",
		        snd_strerror(rc));
	}

	/* Prepare in audioRead() for the first time. */
	sound.reprepare = 1;
}

static ALCboolean alsa_reset_playback(ALCdevice *device)
{
    alsa_data *data = (alsa_data*)device->ExtraData;
    snd_pcm_uframes_t periodSizeInFrames;
    unsigned int periodLen, bufferLen;
    snd_pcm_sw_params_t *sp = NULL;
    snd_pcm_hw_params_t *hp = NULL;
    snd_pcm_access_t access;
    snd_pcm_format_t format;
    unsigned int periods;
    unsigned int rate;
    const char *funcerr;
    int allowmmap;
    int err;

    format = -1;
    switch(device->FmtType)
    {
        case DevFmtByte:
            format = SND_PCM_FORMAT_S8;
            break;
        case DevFmtUByte:
            format = SND_PCM_FORMAT_U8;
            break;
        case DevFmtShort:
            format = SND_PCM_FORMAT_S16;
            break;
        case DevFmtUShort:
            format = SND_PCM_FORMAT_U16;
            break;
        case DevFmtInt:
            format = SND_PCM_FORMAT_S32;
            break;
        case DevFmtUInt:
            format = SND_PCM_FORMAT_U32;
            break;
        case DevFmtFloat:
            format = SND_PCM_FORMAT_FLOAT;
            break;
    }

    allowmmap = GetConfigValueBool("alsa", "mmap", 1);
    periods = device->NumUpdates;
    periodLen = (ALuint64)device->UpdateSize * 1000000 / device->Frequency;
    bufferLen = periodLen * periods;
    rate = device->Frequency;

    snd_pcm_hw_params_malloc(&hp);
#define CHECK(x) if((funcerr=#x),(err=(x)) < 0) goto error
    CHECK(snd_pcm_hw_params_any(data->pcmHandle, hp));
    /* set interleaved access */
    if(!allowmmap || snd_pcm_hw_params_set_access(data->pcmHandle, hp, SND_PCM_ACCESS_MMAP_INTERLEAVED) < 0)
    {
        if(periods > 2)
        {
            periods--;
            bufferLen = periodLen * periods;
        }
        CHECK(snd_pcm_hw_params_set_access(data->pcmHandle, hp, SND_PCM_ACCESS_RW_INTERLEAVED));
    }
    /* test and set format (implicitly sets sample bits) */
    if(snd_pcm_hw_params_test_format(data->pcmHandle, hp, format) < 0)
    {
        static const struct {
            snd_pcm_format_t format;
            enum DevFmtType fmttype;
        } formatlist[] = {
            { SND_PCM_FORMAT_FLOAT, DevFmtFloat  },
            { SND_PCM_FORMAT_S32,   DevFmtInt    },
            { SND_PCM_FORMAT_U32,   DevFmtUInt   },
            { SND_PCM_FORMAT_S16,   DevFmtShort  },
            { SND_PCM_FORMAT_U16,   DevFmtUShort },
            { SND_PCM_FORMAT_S8,    DevFmtByte   },
            { SND_PCM_FORMAT_U8,    DevFmtUByte  },
        };
        size_t k;

        for(k = 0;k < COUNTOF(formatlist);k++)
        {
            format = formatlist[k].format;
            if(snd_pcm_hw_params_test_format(data->pcmHandle, hp, format) >= 0)
            {
                device->FmtType = formatlist[k].fmttype;
                break;
            }
        }
    }
    CHECK(snd_pcm_hw_params_set_format(data->pcmHandle, hp, format));
    /* test and set channels (implicitly sets frame bits) */
    if(snd_pcm_hw_params_test_channels(data->pcmHandle, hp, ChannelsFromDevFmt(device->FmtChans)) < 0)
    {
        static const enum DevFmtChannels channellist[] = {
            DevFmtStereo,
            DevFmtQuad,
            DevFmtX51,
            DevFmtX71,
            DevFmtMono,
        };
        size_t k;

//.........这里部分代码省略.........

static ALCenum alsa_open_capture(ALCdevice *Device, const ALCchar *deviceName)
{
    const char *driver = NULL;
    snd_pcm_hw_params_t *hp;
    snd_pcm_uframes_t bufferSizeInFrames;
    snd_pcm_uframes_t periodSizeInFrames;
    ALboolean needring = AL_FALSE;
    snd_pcm_format_t format;
    const char *funcerr;
    alsa_data *data;
    int err;

    if(deviceName)
    {
        size_t idx;

        if(!allCaptureDevNameMap)
            allCaptureDevNameMap = probe_devices(SND_PCM_STREAM_CAPTURE, &numCaptureDevNames);

        for(idx = 0;idx < numCaptureDevNames;idx++)
        {
            if(strcmp(deviceName, allCaptureDevNameMap[idx].name) == 0)
            {
                driver = allCaptureDevNameMap[idx].device;
                break;
            }
        }
        if(idx == numCaptureDevNames)
            return ALC_INVALID_VALUE;
    }
    else
    {
        deviceName = alsaDevice;
        driver = GetConfigValue("alsa", "capture", "default");
    }

    data = (alsa_data*)calloc(1, sizeof(alsa_data));

    err = snd_pcm_open(&data->pcmHandle, driver, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK);
    if(err < 0)
    {
        ERR("Could not open capture device '%s': %s\n", driver, snd_strerror(err));
        free(data);
        return ALC_INVALID_VALUE;
    }

    format = -1;
    switch(Device->FmtType)
    {
        case DevFmtByte:
            format = SND_PCM_FORMAT_S8;
            break;
        case DevFmtUByte:
            format = SND_PCM_FORMAT_U8;
            break;
        case DevFmtShort:
            format = SND_PCM_FORMAT_S16;
            break;
        case DevFmtUShort:
            format = SND_PCM_FORMAT_U16;
            break;
        case DevFmtInt:
            format = SND_PCM_FORMAT_S32;
            break;
        case DevFmtUInt:
            format = SND_PCM_FORMAT_U32;
            break;
        case DevFmtFloat:
            format = SND_PCM_FORMAT_FLOAT;
            break;
    }

    funcerr = NULL;
    bufferSizeInFrames = maxu(Device->UpdateSize*Device->NumUpdates,
                              100*Device->Frequency/1000);
    periodSizeInFrames = minu(bufferSizeInFrames, 25*Device->Frequency/1000);

    snd_pcm_hw_params_malloc(&hp);
#define CHECK(x) if((funcerr=#x),(err=(x)) < 0) goto error
    CHECK(snd_pcm_hw_params_any(data->pcmHandle, hp));
    /* set interleaved access */
    CHECK(snd_pcm_hw_params_set_access(data->pcmHandle, hp, SND_PCM_ACCESS_RW_INTERLEAVED));
    /* set format (implicitly sets sample bits) */
    CHECK(snd_pcm_hw_params_set_format(data->pcmHandle, hp, format));
    /* set channels (implicitly sets frame bits) */
    CHECK(snd_pcm_hw_params_set_channels(data->pcmHandle, hp, ChannelsFromDevFmt(Device->FmtChans)));
    /* set rate (implicitly constrains period/buffer parameters) */
    CHECK(snd_pcm_hw_params_set_rate(data->pcmHandle, hp, Device->Frequency, 0));
    /* set buffer size in frame units (implicitly sets period size/bytes/time and buffer time/bytes) */
    if(snd_pcm_hw_params_set_buffer_size_min(data->pcmHandle, hp, &bufferSizeInFrames) < 0)
    {
        TRACE("Buffer too large, using intermediate ring buffer\n");
        needring = AL_TRUE;
        CHECK(snd_pcm_hw_params_set_buffer_size_near(data->pcmHandle, hp, &bufferSizeInFrames));
    }
    /* set buffer size in frame units (implicitly sets period size/bytes/time and buffer time/bytes) */
    CHECK(snd_pcm_hw_params_set_period_size_near(data->pcmHandle, hp, &periodSizeInFrames, NULL));
    /* install and prepare hardware configuration */
    CHECK(snd_pcm_hw_params(data->pcmHandle, hp));
    /* retrieve configuration info */
//.........这里部分代码省略.........

static int
quh_alsa_config (st_quh_nfo_t *file)
{
  (void) file;
#if 0
  snd_pcm_hw_params_t *hw_params;
  snd_pcm_format_t format;
  int rate = 0;
  
  if (snd_pcm_hw_params_malloc (&hw_params) < 0)
    return -1;

  if (snd_pcm_hw_params_any (handle, hw_params) < 0)
    {
      snd_pcm_hw_params_free (hw_params);
      return -1;
    }

  if (snd_pcm_hw_params_set_access (handle, hw_params,
                                    SND_PCM_ACCESS_RW_INTERLEAVED) < 0)
    {
      snd_pcm_hw_params_free (hw_params);
      return -1;
    }

  switch (file->size)
    {
    case 1:
      format = SND_PCM_FORMAT_S8;
      break;
    case 2:
      format = SND_PCM_FORMAT_S16;
      break;
    case 3:
      format = SND_PCM_FORMAT_S24;
      break;
    default:
      format = SND_PCM_FORMAT_S16;
      break;
    }

  if (snd_pcm_hw_params_set_format (handle, hw_params, format) < 0)
    {
      snd_pcm_hw_params_free (hw_params);
      return -1;
    }

  rate = file->rate;
  if (snd_pcm_hw_params_set_rate_near (handle, hw_params, rate, 0) < 0)
    {
      snd_pcm_hw_params_free (hw_params);
      return -1;
    }

  if ((float) rate * 1.05 < file->rate || (float) rate * 0.95 > file->rate)
    {
      snd_pcm_hw_params_free (hw_params);
      return -1;
    }

  if (snd_pcm_hw_params_set_channels (handle, hw_params, file->channels) < 0)
    {
      snd_pcm_hw_params_free (hw_params);
      return -1;
    }

  if (snd_pcm_hw_params (handle, hw_params) < 0)
    {
      snd_pcm_hw_params_free (hw_params);
      return -1;
    }

  snd_pcm_hw_params_free (hw_params);
#endif
  return 0;
}

av_cold int ff_alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode,
                         unsigned int *sample_rate,
                         int channels, enum CodecID *codec_id)
{
    AlsaData *s = ctx->priv_data;
    const char *audio_device;
    int res, flags = 0;
    snd_pcm_format_t format;
    snd_pcm_t *h;
    snd_pcm_hw_params_t *hw_params;
    snd_pcm_uframes_t buffer_size, period_size;
    int64_t layout = ctx->streams[0]->codec->channel_layout;

    if (ctx->filename[0] == 0) audio_device = "default";
    else                       audio_device = ctx->filename;

    if (*codec_id == CODEC_ID_NONE)
        *codec_id = DEFAULT_CODEC_ID;
    format = codec_id_to_pcm_format(*codec_id);
    if (format == SND_PCM_FORMAT_UNKNOWN) {
        av_log(ctx, AV_LOG_ERROR, "sample format 0x%04x is not supported\n", *codec_id);
        return AVERROR(ENOSYS);
    }
    s->frame_size = av_get_bits_per_sample(*codec_id) / 8 * channels;

    if (ctx->flags & AVFMT_FLAG_NONBLOCK) {
        flags = SND_PCM_NONBLOCK;
    }
    res = snd_pcm_open(&h, audio_device, mode, flags);
    if (res < 0) {
        av_log(ctx, AV_LOG_ERROR, "cannot open audio device %s (%s)\n",
               audio_device, snd_strerror(res));
        return AVERROR(EIO);
    }

    res = snd_pcm_hw_params_malloc(&hw_params);
    if (res < 0) {
        av_log(ctx, AV_LOG_ERROR, "cannot allocate hardware parameter structure (%s)\n",
               snd_strerror(res));
        goto fail1;
    }

    res = snd_pcm_hw_params_any(h, hw_params);
    if (res < 0) {
        av_log(ctx, AV_LOG_ERROR, "cannot initialize hardware parameter structure (%s)\n",
               snd_strerror(res));
        goto fail;
    }

    res = snd_pcm_hw_params_set_access(h, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
    if (res < 0) {
        av_log(ctx, AV_LOG_ERROR, "cannot set access type (%s)\n",
               snd_strerror(res));
        goto fail;
    }

    res = snd_pcm_hw_params_set_format(h, hw_params, format);
    if (res < 0) {
        av_log(ctx, AV_LOG_ERROR, "cannot set sample format 0x%04x %d (%s)\n",
               *codec_id, format, snd_strerror(res));
        goto fail;
    }

    res = snd_pcm_hw_params_set_rate_near(h, hw_params, sample_rate, 0);
    if (res < 0) {
        av_log(ctx, AV_LOG_ERROR, "cannot set sample rate (%s)\n",
               snd_strerror(res));
        goto fail;
    }

    res = snd_pcm_hw_params_set_channels(h, hw_params, channels);
    if (res < 0) {
        av_log(ctx, AV_LOG_ERROR, "cannot set channel count to %d (%s)\n",
               channels, snd_strerror(res));
        goto fail;
    }

    snd_pcm_hw_params_get_buffer_size_max(hw_params, &buffer_size);
    buffer_size = FFMIN(buffer_size, ALSA_BUFFER_SIZE_MAX);
    /* TODO: maybe use ctx->max_picture_buffer somehow */
    res = snd_pcm_hw_params_set_buffer_size_near(h, hw_params, &buffer_size);
    if (res < 0) {
        av_log(ctx, AV_LOG_ERROR, "cannot set ALSA buffer size (%s)\n",
               snd_strerror(res));
        goto fail;
    }

    snd_pcm_hw_params_get_period_size_min(hw_params, &period_size, NULL);
    if (!period_size)
        period_size = buffer_size / 4;
    res = snd_pcm_hw_params_set_period_size_near(h, hw_params, &period_size, NULL);
    if (res < 0) {
        av_log(ctx, AV_LOG_ERROR, "cannot set ALSA period size (%s)\n",
               snd_strerror(res));
        goto fail;
    }
    s->period_size = period_size;

    res = snd_pcm_hw_params(h, hw_params);
    if (res < 0) {
//.........这里部分代码省略.........

alsa_dev_t* alsa_open(char *name)
{
    alsa_dev_t *alsa_dev = NULL;
    snd_pcm_hw_params_t *params = NULL;
    uint32_t val = 0;
    int32_t rc = 0;
    int32_t dir = 0;

    alsa_dev = malloc(sizeof(alsa_dev_t));
    if (!alsa_dev)
    {
        syslog(LOG_ERR, "Failed to create audio device: Out of memory");
        exit(EXIT_FAILURE);
    }

    /* Open PCM device for recording (capture). */
    rc = snd_pcm_open(&alsa_dev->snd_pcm, name, SND_PCM_STREAM_CAPTURE, 0);
    if (rc < 0)
    {
        syslog(LOG_ERR, "Cannot open pcm device: %s", snd_strerror(rc));
        exit(EXIT_FAILURE);
    }

    alsa_dev->name = name;

    /* Allocate a hardware parameters object. */
    snd_pcm_hw_params_alloca(&params);

    /* Fill it in with default values. */
    snd_pcm_hw_params_any(alsa_dev->snd_pcm, params);

    /* Set the desired hardware parameters. */

    /* Interleaved mode */
    snd_pcm_hw_params_set_access(alsa_dev->snd_pcm, params, 
                                 SND_PCM_ACCESS_RW_INTERLEAVED);

    /* Signed 16-bit little-endian format */
    snd_pcm_hw_params_set_format(alsa_dev->snd_pcm, params, 
                                 SND_PCM_FORMAT_S16_LE);

    /* Two channels (stereo) */
    snd_pcm_hw_params_set_channels(alsa_dev->snd_pcm, params, AUDIO_CHANNEL_NUM);

    /* 44100 bits/second sampling rate (CD quality) */
    val = AUDIO_SAMPLE_RATE;
    snd_pcm_hw_params_set_rate_near(alsa_dev->snd_pcm, params, &val, &dir);

    alsa_dev->frames = 1152;
    snd_pcm_hw_params_set_period_size_near(alsa_dev->snd_pcm, params, 
                                           &alsa_dev->frames, &dir);

    /* Write the parameters to the driver */
    rc = snd_pcm_hw_params(alsa_dev->snd_pcm, params);
    if (rc < 0)
    {
        syslog(LOG_ERR, "Cannot set hw parameters: %s", snd_strerror(rc));
        exit(EXIT_FAILURE);
    }

    /* Use a buffer large enough to hold one period */
    snd_pcm_hw_params_get_period_size(params, &alsa_dev->frames, &dir);

    /* We want to loop for 5 seconds */
    snd_pcm_hw_params_get_period_time(params, &val, &dir);

    snd_pcm_prepare(alsa_dev->snd_pcm);

    snd_pcm_start(alsa_dev->snd_pcm);

    return alsa_dev;
}

int main()
{
	int fp;
  unsigned int pcm, tmp, dir;
  int buff_size;
	
  long loops;
  int rc;
  int size;
  snd_pcm_t *handle;
  snd_pcm_hw_params_t *params;
  unsigned int val;
  
  snd_pcm_uframes_t frames;
  char *buff;
  int rate, channels, seconds;

  /* Open PCM device for recording (capture). */
  rc = snd_pcm_open(&handle, "default", SND_PCM_STREAM_CAPTURE, 0);

  if (rc < 0) 
  {
    fprintf(stderr,"unable to open pcm device: %s\n",   snd_strerror(rc));
    exit(1);
  }

  /* Allocate a hardware parameters object. */
  snd_pcm_hw_params_alloca(&params);

  /* Fill it in with default values. */
  snd_pcm_hw_params_any(handle, params);

  /* Set the desired hardware parameters. */

  /* Interleaved mode */
  snd_pcm_hw_params_set_access(handle, params,  SND_PCM_ACCESS_RW_INTERLEAVED);

  /* Signed 16-bit little-endian format */
  snd_pcm_hw_params_set_format(handle, params,SND_PCM_FORMAT_U8);

  /* Two channels (stereo) */
  snd_pcm_hw_params_set_channels(handle, params, 1);

  /* 44100 bits/second sampling rate (CD quality) */
  val = 8000;
  
  snd_pcm_hw_params_set_rate_near(handle, params,  &val, &dir);

  /* Set period size to 32 frames. */
  frames = 32;
  snd_pcm_hw_params_set_period_size_near(handle,params, &frames, &dir);

  /* Write the parameters to the driver */
  rc = snd_pcm_hw_params(handle, params);
  if (rc < 0) 
  {
    fprintf(stderr,"unable to set hw parameters: %s\n", snd_strerror(rc));
    exit(1);
  }

  /* Use a buff large enough to hold one period */

  snd_pcm_hw_params_get_period_size(params,  &frames, &dir);
  size = frames * 4; /* 2 bytes/sample, 2 channels */
  buff = (char *) malloc(size);

  /* We want to loop for 5 seconds */

  snd_pcm_hw_params_get_period_time(params, &val, &dir);
  loops = 50000000 / val;

	fp=open("222.wav",O_WRONLY);
  while (loops > 0) 
  {
		loops--;
		bzero(buff,sizeof(buff));
		rc = snd_pcm_readi(handle, buff, frames);
		write(fp,buff,128);
		if (rc == -EPIPE) 
		{
		  /* EPIPE means overrun */
		  fprintf(stderr, "overrun occurred\n");
		  snd_pcm_prepare(handle);
		} else if (rc < 0) 
		{
		  fprintf(stderr,"error from read: %s\n",  snd_strerror(rc));
		} else if (rc != (int)frames) 
		{
		  fprintf(stderr, "short read, read %d frames\n", rc);
		}
		
		/*rc = write(1, buff, size);
		if (rc != size)
		  fprintf(stderr,"short write: wrote %d bytes\n", rc);
	*/
  }

  snd_pcm_drain(handle);
  snd_pcm_close(handle);
  
//.........这里部分代码省略.........

// We're making these functions static - there's no need to pollute the global namespace
static void alsa_init(audio_backend_pull_callback_t pull_callback)
{
	snd_pcm_hw_params_t *params;
	unsigned int val;
	int dir=-1;

	string device = cfgLoadStr("alsa", "device", "");

	int rc = -1;
	if (device == "" || device == "auto")
	{
		printf("ALSA: trying to determine audio device\n");

		// trying default device
		device = "default";
		rc = snd_pcm_open(&handle, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0);

		// "default" didn't work, try first device
		if (rc < 0)
		{
			device = "plughw:0,0,0";
			rc = snd_pcm_open(&handle, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0);

			if (rc < 0)
			{
				device = "plughw:0,0";
				rc = snd_pcm_open(&handle, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0);
			}
		}

		// first didn't work, try second
		if (rc < 0)
		{
			device = "plughw:1,0";
			rc = snd_pcm_open(&handle, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0);
		}

		// try pulse audio backend
		if (rc < 0)
		{
			device = "pulse";
			rc = snd_pcm_open(&handle, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0);
		}

		if (rc < 0)
			printf("ALSA: unable to automatically determine audio device.\n");
	}
	else {
		rc = snd_pcm_open(&handle, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0);
	}

	if (rc < 0)
	{
		fprintf(stderr, "ALSA: unable to open PCM device %s: %s\n", device.c_str(), snd_strerror(rc));
		return;
	}

	printf("ALSA: Successfully initialized \"%s\"\n", device.c_str());

	/* Allocate a hardware parameters object. */
	snd_pcm_hw_params_alloca(&params);

	/* Fill it in with default values. */
	rc=snd_pcm_hw_params_any(handle, params);
	if (rc < 0)
	{
		fprintf(stderr, "ALSA: Error:snd_pcm_hw_params_any %s\n", snd_strerror(rc));
		return;
	}

	/* Set the desired hardware parameters. */

	/* Interleaved mode */
	rc=snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
	if (rc < 0)
	{
		fprintf(stderr, "ALSA: Error:snd_pcm_hw_params_set_access %s\n", snd_strerror(rc));
		return;
	}

	/* Signed 16-bit little-endian format */
	rc=snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE);
	if (rc < 0)
	{
		fprintf(stderr, "ALSA: Error:snd_pcm_hw_params_set_format %s\n", snd_strerror(rc));
		return;
	}

	/* Two channels (