C++ snd_pcm_hw_params_free函数代码示例

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本文整理汇总了C++中snd_pcm_hw_params_free函数的典型用法代码示例。如果您正苦于以下问题：C++ snd_pcm_hw_params_free函数的具体用法？C++ snd_pcm_hw_params_free怎么用？C++ snd_pcm_hw_params_free使用的例子？那么恭喜您, 这里精选的函数代码示例或许可以为您提供帮助。

在下文中一共展示了snd_pcm_hw_params_free函数的20个代码示例，这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞，您的评价将有助于我们的系统推荐出更棒的C++代码示例。

示例1: quh_alsa_config

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;
}

开发者ID:BackupTheBerlios，项目名称:quh，代码行数:76，代码来源:alsa.c

示例2: detect_pcm

void *capture_thread(void *data) {
  AudioCapture *capture = (AudioCapture *)data;
  // enif_keep_resource(capture);
  
  detect_pcm(capture);
  
  set_volume(capture, 100);
  
  snd_pcm_open(&capture->handle, capture->pcm_name, SND_PCM_STREAM_CAPTURE, 0);
  if(!capture->handle) {
    fprintf(stderr, "No PCM!!\r\n");
    exit(1);
  }
  snd_pcm_hw_params_t *hw_params;
  snd_pcm_hw_params_malloc(&hw_params);
  if(!hw_params) {
    fprintf(stderr, "Damn!! No hw_params\r\n");
    exit(1);
  }
  
  snd_pcm_hw_params_any(capture->handle, hw_params);
  snd_pcm_hw_params_set_access(capture->handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
  snd_pcm_hw_params_set_format(capture->handle, hw_params, SND_PCM_FORMAT_S16_LE);
  snd_pcm_hw_params_set_rate_near(capture->handle, hw_params, &capture->sample_rate, 0);
  snd_pcm_hw_params_set_channels(capture->handle, hw_params, capture->channels);
  
  // int buffer_size = 16384;
  // int period_time = 32000;
  // int period_size = 1024;
  // snd_pcm_hw_params_set_period_time_near(capture->handle, hw_params, &period_time, 0);
  // snd_pcm_hw_params_set_period_size_near(capture->handle, hw_params, &period_size, 0);
  //  snd_pcm_hw_params_set_buffer_size_near(capture->handle, hw_params, &buffer_size);
  //  
	
  capture->last_dts = 0;
  
  fprintf(stderr, "Setting params: %p, %p\r\n", capture->handle, hw_params);
  
  snd_pcm_hw_params(capture->handle, hw_params);
	snd_pcm_hw_params_free(hw_params);
  snd_pcm_prepare(capture->handle);
  
  snd_output_t *log;
  snd_output_stdio_attach(&log, stderr, 0);
  snd_pcm_hw_params_dump(hw_params, log);
  
  fprintf(stderr, "Started capture\r\n");
  char *buffer = (char *)malloc(8192);
  
  // char *ptr = buffer;
  int size = 0;
  
  while(capture->thread_started) {
    int r = snd_pcm_readi(capture->handle, buffer + size, 1024);
    size += r*2*capture->channels;
    if(size < capture->frame_size) {
      continue;
    }
    
    ErlNifEnv* env = enif_alloc_env();
    ErlNifBinary frame;
    
    enif_alloc_binary(capture->frame_size, &frame);
    memmove(frame.data, buffer, frame.size);
    size -= frame.size;
    memmove(buffer, buffer + frame.size, size);
    
    ErlNifUInt64 dts = (uint64_t)capture->counter*1024ULL*1000 / capture->sample_rate;
    
    // fprintf(stderr, "A: %d -> %d\r\n", capture->counter, dts);
    
    if(capture->last_dts > dts) {
      fprintf(stderr, "Achtung! ALSA audio jump: %u, %u, %u\r\n", (unsigned)capture->counter, (unsigned)capture->last_dts, (unsigned)dts);
    }
    capture->last_dts = dts;
    enif_send(NULL, &capture->owner_pid, env, 
      enif_make_tuple4(env,
        enif_make_atom(env, "alsa"),
        enif_make_resource(env, capture),
        enif_make_uint64(env, dts),
        enif_make_binary(env, &frame)
      )
    );
    
    enif_release_binary(&frame);
    
    enif_free_env(env);
    capture->counter++;
  }
  fprintf(stderr, "Capture thread stopping\r\n");
  // enif_release_resource(capture);
  snd_pcm_close(capture->handle);
  return 0;
}

开发者ID:DEGAUSSE，项目名称:alsa，代码行数:94，代码来源:alsa.c

示例3: calloc

static void *alsa_thread_init(const char *device,
      unsigned rate, unsigned latency)
{
   snd_pcm_uframes_t buffer_size;
   snd_pcm_format_t format;
   snd_pcm_hw_params_t *params    = NULL;
   snd_pcm_sw_params_t *sw_params = NULL;
   const char *alsa_dev           = device ? device : "default";
   unsigned latency_usec          = latency * 1000 / 2;
   unsigned channels              = 2;
   unsigned periods               = 4;
   alsa_thread_t            *alsa = (alsa_thread_t*)
      calloc(1, sizeof(alsa_thread_t));

   if (!alsa)
      return NULL;

   TRY_ALSA(snd_pcm_open(&alsa->pcm, alsa_dev, SND_PCM_STREAM_PLAYBACK, 0));

   TRY_ALSA(snd_pcm_hw_params_malloc(&params));
   alsa->has_float = alsathread_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));

   /* Shouldn't have to bother with this, 
    * but some drivers are apparently broken. */
   if (snd_pcm_hw_params_get_period_size(params, &alsa->period_frames, NULL))
      snd_pcm_hw_params_get_period_size_min(
            params, &alsa->period_frames, NULL);
   RARCH_LOG("ALSA: Period size: %d frames\n", (int)alsa->period_frames);
   if (snd_pcm_hw_params_get_buffer_size(params, &buffer_size))
      snd_pcm_hw_params_get_buffer_size_max(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->period_size = snd_pcm_frames_to_bytes(alsa->pcm, alsa->period_frames);

   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);

   alsa->fifo_lock = slock_new();
   alsa->cond_lock = slock_new();
   alsa->cond = scond_new();
   alsa->buffer = fifo_new(alsa->buffer_size);
   if (!alsa->fifo_lock || !alsa->cond_lock || !alsa->cond || !alsa->buffer)
      goto error;

   alsa->worker_thread = sthread_create(alsa_worker_thread, alsa);
   if (!alsa->worker_thread)
   {
      RARCH_ERR("error initializing worker thread");
      goto error;
   }

   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);

   alsa_thread_free(alsa);

   return NULL;
}

开发者ID:Ezio-PS，项目名称:RetroArch，代码行数:87，代码来源:alsathread.c

示例4: alsa_configure

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.........

开发者ID:mmaruska，项目名称:emacs，代码行数:101，代码来源:sound.c

示例5: set_hwparams


//.........这里部分代码省略.........
            GST_ELEMENT_WARNING (alsa, RESOURCE, SETTINGS, (NULL),
                                 ("Unable to set period time %i for playback: %s",
                                  period_time, snd_strerror (err)));
            /* disable period_time the next round */
            period_time = -1;
            goto retry;
        }
        GST_DEBUG_OBJECT (alsa, "period time %u", period_time);
    }

    /* Set buffer size and period size manually for SPDIF */
    if (G_UNLIKELY (alsa->iec958)) {
        snd_pcm_uframes_t buffer_size = SPDIF_BUFFER_SIZE;
        snd_pcm_uframes_t period_size = SPDIF_PERIOD_SIZE;

        CHECK (snd_pcm_hw_params_set_buffer_size_near (alsa->handle, params,
                &buffer_size), buffer_size);
        CHECK (snd_pcm_hw_params_set_period_size_near (alsa->handle, params,
                &period_size, NULL), period_size);
    }

    /* write the parameters to device */
    CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);

    /* now get the configured values */
    CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size),
           buffer_size);
    CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size, NULL),
           period_size);

    GST_DEBUG_OBJECT (alsa, "buffer size %lu, period size %lu", alsa->buffer_size,
                      alsa->period_size);

    snd_pcm_hw_params_free (params);
    return 0;

    /* ERRORS */
no_config:
    {
        GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
                           ("Broken configuration for playback: no configurations available: %s",
                            snd_strerror (err)));
        snd_pcm_hw_params_free (params);
        return err;
    }
wrong_access:
    {
        GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
                           ("Access type not available for playback: %s", snd_strerror (err)));
        snd_pcm_hw_params_free (params);
        return err;
    }
no_sample_format:
    {
        GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
                           ("Sample format not available for playback: %s", snd_strerror (err)));
        snd_pcm_hw_params_free (params);
        return err;
    }
no_channels:
    {
        gchar *msg = NULL;

        if ((alsa->channels) == 1)
            msg = g_strdup (_("Could not open device for playback in mono mode."));
        if ((alsa->channels) == 2)

开发者ID:pli3，项目名称:gst-plugins-base，代码行数:67，代码来源:gstalsasink.c

示例6: snd_pcm_hw_params_free

AudioSourceNix::~AudioSourceNix()
{
    snd_pcm_hw_params_free(params);
}

开发者ID:1981khj，项目名称:rainbow，代码行数:4，代码来源:AudioSourceNix.cpp

示例7: alsa_open

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.........

开发者ID:5in4，项目名称:libsox.dll，代码行数:101，代码来源:sndfile-play.c

示例8: initAlsa

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);
}

开发者ID:ngreatorex，项目名称:acarsdec，代码行数:73，代码来源:alsa.c

示例9: alsa_set_format

static RD_BOOL
alsa_set_format(snd_pcm_t * pcm, RD_WAVEFORMATEX * pwfx)
{
	snd_pcm_hw_params_t *hwparams = NULL;
	int err;
	unsigned int buffertime;
	short samplewidth;
	int audiochannels;
	unsigned int rate;

	samplewidth = pwfx->wBitsPerSample / 8;

	if ((err = snd_pcm_hw_params_malloc(&hwparams)) < 0)
	{
		error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
		return False;
	}

	if ((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0)
	{
		error("snd_pcm_hw_params_any: %s\n", snd_strerror(err));
		return False;
	}

	if ((err = snd_pcm_hw_params_set_access(pcm, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
	{
		error("snd_pcm_hw_params_set_access: %s\n", snd_strerror(err));
		return False;
	}

	if (pwfx->wBitsPerSample == 16)
	{
		if ((err = snd_pcm_hw_params_set_format(pcm, hwparams, SND_PCM_FORMAT_S16_LE)) < 0)
		{
			error("snd_pcm_hw_params_set_format: %s\n", snd_strerror(err));
			return False;
		}
	}
	else
	{
		if ((err = snd_pcm_hw_params_set_format(pcm, hwparams, SND_PCM_FORMAT_S8)) < 0)
		{
			error("snd_pcm_hw_params_set_format: %s\n", snd_strerror(err));
			return False;
		}
	}

#if 0
	if ((err = snd_pcm_hw_params_set_rate_resample(pcm, hwparams, 1)) < 0)
	{
		error("snd_pcm_hw_params_set_rate_resample: %s\n", snd_strerror(err));
		return False;
	}
#endif

	rate = pwfx->nSamplesPerSec;
	if ((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0)
	{
		error("snd_pcm_hw_params_set_rate_near: %s\n", snd_strerror(err));
		return False;
	}

	audiochannels = pwfx->nChannels;
	if ((err = snd_pcm_hw_params_set_channels(pcm, hwparams, pwfx->nChannels)) < 0)
	{
		error("snd_pcm_hw_params_set_channels: %s\n", snd_strerror(err));
		return False;
	}


	buffertime = 500000;	/* microseconds */
	if ((err = snd_pcm_hw_params_set_buffer_time_near(pcm, hwparams, &buffertime, 0)) < 0)
	{
		error("snd_pcm_hw_params_set_buffer_time_near: %s\n", snd_strerror(err));
		return False;
	}

	if ((err = snd_pcm_hw_params(pcm, hwparams)) < 0)
	{
		error("snd_pcm_hw_params: %s\n", snd_strerror(err));
		return False;
	}

	snd_pcm_hw_params_free(hwparams);

	if ((err = snd_pcm_prepare(pcm)) < 0)
	{
		error("snd_pcm_prepare: %s\n", snd_strerror(err));
		return False;
	}

	reopened = True;

	return True;
}

开发者ID:AmesianX，项目名称:rdesktop-fuzzer，代码行数:95，代码来源:rdpsnd_alsa.c

示例10: main

// alsa	      
	main (int argc, char *argv[])
	{
		int i;
		int err;
		short buf[128];
		snd_pcm_t *playback_handle;
		snd_pcm_hw_params_t *hw_params;
	
		if ((err = snd_pcm_open (&playback_handle, argv[1], SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
			fprintf (stderr, "cannot open audio device %s (%s)\n", 
				 argv[1],
				 snd_strerror (err));
			exit (1);
		}
		   
		if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) {
			fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n",
				 snd_strerror (err));
			exit (1);
		}
				 
		if ((err = snd_pcm_hw_params_any (playback_handle, hw_params)) < 0) {
			fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n",
				 snd_strerror (err));
			exit (1);
		}
	
		if ((err = snd_pcm_hw_params_set_access (playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
			fprintf (stderr, "cannot set access type (%s)\n",
				 snd_strerror (err));
			exit (1);
		}
	
		if ((err = snd_pcm_hw_params_set_format (playback_handle, hw_params, SND_PCM_FORMAT_S16_LE)) < 0) {
			fprintf (stderr, "cannot set sample format (%s)\n",
				 snd_strerror (err));
			exit (1);
		}
	
		if ((err = snd_pcm_hw_params_set_rate_near (playback_handle, hw_params, 44100, 0)) < 0) {
			fprintf (stderr, "cannot set sample rate (%s)\n",
				 snd_strerror (err));
			exit (1);
		}
	
		if ((err = snd_pcm_hw_params_set_channels (playback_handle, hw_params, 2)) < 0) {
			fprintf (stderr, "cannot set channel count (%s)\n",
				 snd_strerror (err));
			exit (1);
		}
	
		if ((err = snd_pcm_hw_params (playback_handle, hw_params)) < 0) {
			fprintf (stderr, "cannot set parameters (%s)\n",
				 snd_strerror (err));
			exit (1);
		}
	
		snd_pcm_hw_params_free (hw_params);
	
		if ((err = snd_pcm_prepare (playback_handle)) < 0) {
			fprintf (stderr, "cannot prepare audio interface for use (%s)\n",
				 snd_strerror (err));
			exit (1);
		}
	
		for (i = 0; i < 10; ++i) {
			if ((err = snd_pcm_writei (playback_handle, buf, 128)) != 128) {
				fprintf (stderr, "write to audio interface failed (%s)\n",
					 snd_strerror (err));
				exit (1);
			}
		}
	
		snd_pcm_close (playback_handle);
		exit (0);
	}

开发者ID:1090310214，项目名称:sound-card，代码行数:77，代码来源:alsa练习1.c

示例11: alsa_open_capture


//.........这里部分代码省略.........
        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 */
    CHECK(snd_pcm_hw_params_get_period_size(hp, &periodSizeInFrames, NULL));
#undef CHECK
    snd_pcm_hw_params_free(hp);
    hp = NULL;

    if(needring)
    {
        data->ring = CreateRingBuffer(FrameSizeFromDevFmt(Device->FmtChans, Device->FmtType),
                                      Device->UpdateSize*Device->NumUpdates);
        if(!data->ring)
        {
            ERR("ring buffer create failed\n");
            goto error2;
        }

        data->size = snd_pcm_frames_to_bytes(data->pcmHandle, periodSizeInFrames);
        data->buffer = malloc(data->size);
        if(!data->buffer)
        {
            ERR("buffer malloc failed\n");
            goto error2;
        }
    }

    Device->DeviceName = strdup(deviceName);

    Device->ExtraData = data;
    return ALC_NO_ERROR;

error:
    ERR("%s failed: %s\n", funcerr, snd_strerror(err));
    if(hp) snd_pcm_hw_params_free(hp);

error2:
    free(data->buffer);
    DestroyRingBuffer(data->ring);
    snd_pcm_close(data->pcmHandle);
    free(data);

    Device->ExtraData = NULL;
    return ALC_INVALID_VALUE;
}

开发者ID:LighFusion，项目名称:surreal，代码行数:101，代码来源:alsa.c

示例12: alsa_reset_playback


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

        for(k = 0;k < COUNTOF(channellist);k++)
        {
            if(snd_pcm_hw_params_test_channels(data->pcmHandle, hp, ChannelsFromDevFmt(channellist[k])) >= 0)
            {
                device->FmtChans = channellist[k];
                break;
            }
        }
    }
    CHECK(snd_pcm_hw_params_set_channels(data->pcmHandle, hp, ChannelsFromDevFmt(device->FmtChans)));
    /* set rate (implicitly constrains period/buffer parameters) */
    if(snd_pcm_hw_params_set_rate_resample(data->pcmHandle, hp, 0) < 0)
        ERR("Failed to disable ALSA resampler\n");
    CHECK(snd_pcm_hw_params_set_rate_near(data->pcmHandle, hp, &rate, NULL));
    /* set buffer time (implicitly constrains period/buffer parameters) */
    CHECK(snd_pcm_hw_params_set_buffer_time_near(data->pcmHandle, hp, &bufferLen, NULL));
    /* set period time (implicitly sets buffer size/bytes/time and period size/bytes) */
    CHECK(snd_pcm_hw_params_set_period_time_near(data->pcmHandle, hp, &periodLen, NULL));
    /* install and prepare hardware configuration */
    CHECK(snd_pcm_hw_params(data->pcmHandle, hp));
    /* retrieve configuration info */
    CHECK(snd_pcm_hw_params_get_access(hp, &access));
    CHECK(snd_pcm_hw_params_get_period_size(hp, &periodSizeInFrames, NULL));
    CHECK(snd_pcm_hw_params_get_periods(hp, &periods, NULL));

    snd_pcm_hw_params_free(hp);
    hp = NULL;
    snd_pcm_sw_params_malloc(&sp);

    CHECK(snd_pcm_sw_params_current(data->pcmHandle, sp));
    CHECK(snd_pcm_sw_params_set_avail_min(data->pcmHandle, sp, periodSizeInFrames));
    CHECK(snd_pcm_sw_params_set_stop_threshold(data->pcmHandle, sp, periodSizeInFrames*periods));
    CHECK(snd_pcm_sw_params(data->pcmHandle, sp));
#undef CHECK
    snd_pcm_sw_params_free(sp);
    sp = NULL;

    /* Increase periods by one, since the temp buffer counts as an extra
     * period */
    if(access == SND_PCM_ACCESS_RW_INTERLEAVED)
        device->NumUpdates = periods+1;
    else
        device->NumUpdates = periods;
    device->UpdateSize = periodSizeInFrames;
    device->Frequency = rate;

    SetDefaultChannelOrder(device);

    return ALC_TRUE;

error:
    ERR("%s failed: %s\n", funcerr, snd_strerror(err));
    if(hp) snd_pcm_hw_params_free(hp);
    if(sp) snd_pcm_sw_params_free(sp);
    return ALC_FALSE;
}

开发者ID:LighFusion，项目名称:surreal，代码行数:101，代码来源:alsa.c

示例13: snd_pcm_hw_params_free

AudioCaptureNode::~AudioCaptureNode()
{
    if(buffer) delete buffer;
    snd_pcm_hw_params_free(hwParams);
    snd_pcm_close(captureHandle);
}

开发者ID:ndoxx，项目名称:BinnoBot，代码行数:6，代码来源:AudioCaptureNode.cpp

示例14: init_audio

static void
init_audio(const char* devname)
{
  snd_pcm_hw_params_t* hwparams = 0;
  snd_pcm_sw_params_t* swparams = 0;
  snd_pcm_uframes_t    bufsize  = 0;
  unsigned int         buftime  = 1000000;
  unsigned int         pertime  = 50000;
  int                  dir      = 0;
  int                  rc;

  if ((rc = snd_output_stdio_attach(&output, stderr, 0)) < 0)
    exit_snd_error(rc, "log output");

  if ((rc = snd_pcm_open(&audio, devname, SND_PCM_STREAM_PLAYBACK, 0)) < 0)
    exit_snd_error(rc, "opening device");

  if ((rc = snd_pcm_hw_params_malloc(&hwparams)) < 0)
    exit_snd_error(rc, "hardware parameters");

  if ((rc = snd_pcm_hw_params_any(audio, hwparams)) < 0)
    exit_snd_error(rc, "hardware parameters");

  if ((rc = snd_pcm_hw_params_set_rate_resample(audio, hwparams, 0)) < 0)
    exit_snd_error(rc, "hardware parameters");

  if ((rc = snd_pcm_hw_params_set_access(audio, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
    exit_snd_error(rc, "access type");

  if ((rc = snd_pcm_hw_params_set_format(audio, hwparams, SND_PCM_FORMAT_S16)) < 0)
    exit_snd_error(rc, "sample format");

  if ((rc = snd_pcm_hw_params_set_channels_near(audio, hwparams, &n_channels)) < 0)
    exit_snd_error(rc, "number of channels");

  if ((rc = snd_pcm_hw_params_set_rate_near(audio, hwparams, &samplerate, &dir)) < 0)
    exit_snd_error(rc, "sample rate");

  if ((rc = snd_pcm_hw_params_set_buffer_time_near(audio, hwparams, &buftime, &dir)) < 0)
    exit_snd_error(rc, "buffer time");

  if ((rc = snd_pcm_hw_params_set_period_time_near(audio, hwparams, &pertime, &dir)) < 0)
    exit_snd_error(rc, "period time");

  if ((rc = snd_pcm_hw_params(audio, hwparams)) < 0)
    exit_snd_error(rc, "applying hardware parameters");

  if ((rc = snd_pcm_hw_params_get_buffer_size(hwparams, &bufsize)) < 0)
    exit_snd_error(rc, "buffer size");

  if ((rc = snd_pcm_hw_params_get_period_size(hwparams, &periodsize, &dir)) < 0)
    exit_snd_error(rc, "period size");

  snd_pcm_hw_params_free(hwparams);

  if ((rc = snd_pcm_sw_params_malloc(&swparams)) < 0)
    exit_snd_error(rc, "software parameters");

  if ((rc = snd_pcm_sw_params_current(audio, swparams)) < 0)
    exit_snd_error(rc, "software parameters");

  if ((rc = snd_pcm_sw_params_set_start_threshold(audio, swparams,
                                                  bufsize / periodsize * periodsize)) < 0)
    exit_snd_error(rc, "start threshold");

  if ((rc = snd_pcm_sw_params(audio, swparams)) < 0)
    exit_snd_error(rc, "applying software parameters");

  snd_pcm_sw_params_free(swparams);

  if ((rc = snd_pcm_prepare(audio)) < 0)
    exit_snd_error(rc, "preparing device");
}

开发者ID:danielkitta，项目名称:kcio，代码行数:73，代码来源:kcplay.c

示例15: ff_alsa_open


//.........这里部分代码省略.........
               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) {
        av_log(ctx, AV_LOG_ERROR, "cannot set parameters (%s)\n",
               snd_strerror(res));
        goto fail;
    }

    snd_pcm_hw_params_free(hw_params);

    if (channels > 2 && layout) {
        if (find_reorder_func(s, *codec_id, layout, mode == SND_PCM_STREAM_PLAYBACK) < 0) {
            char name[128];
            av_get_channel_layout_string(name, sizeof(name), channels, layout);
            av_log(ctx, AV_LOG_WARNING, "ALSA channel layout unknown or unimplemented for %s %s.\n",
                   name, mode == SND_PCM_STREAM_PLAYBACK ? "playback" : "capture");
        }
        if (s->reorder_func) {
            s->reorder_buf_size = buffer_size;
            s->reorder_buf = av_malloc(s->reorder_buf_size * s->frame_size);
            if (!s->reorder_buf)
                goto fail1;
        }
    }

    s->h = h;
    return 0;

fail:
    snd_pcm_hw_params_free(hw_params);
fail1:
    snd_pcm_close(h);
    return AVERROR(EIO);
}

开发者ID:elnormous，项目名称:libav，代码行数:101，代码来源:alsa.c

示例16: ff_alsa_open


//.........这里部分代码省略.........
    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_IO;
    }

    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

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C++ snd_pcm_hw_params_free函数代码示例

示例1: quh_alsa_config

示例2: detect_pcm

示例3: calloc

示例4: alsa_configure

示例5: set_hwparams

示例6: snd_pcm_hw_params_free

示例7: alsa_open

示例8: initAlsa

示例9: alsa_set_format

示例10: main

示例11: alsa_open_capture

示例12: alsa_reset_playback

示例13: snd_pcm_hw_params_free

示例14: init_audio

示例15: ff_alsa_open

示例16: ff_alsa_open

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