C++ GST_AUDIO_INFO_CHANNELS函数代码示例

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本文整理汇总了C++中GST_AUDIO_INFO_CHANNELS函数的典型用法代码示例。如果您正苦于以下问题：C++ GST_AUDIO_INFO_CHANNELS函数的具体用法？C++ GST_AUDIO_INFO_CHANNELS怎么用？C++ GST_AUDIO_INFO_CHANNELS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GST_AUDIO_INFO_CHANNELS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: gst_audio_info_is_equal

/**
 * gst_audio_info_is_equal:
 * @info: a #GstAudioInfo
 * @other: a #GstAudioInfo
 *
 * Compares two #GstAudioInfo and returns whether they are equal or not
 *
 * Returns: %TRUE if @info and @other are equal, else %FALSE.
 *
 * Since: 1.2
 *
 */
gboolean
gst_audio_info_is_equal (const GstAudioInfo * info, const GstAudioInfo * other)
{
  if (info == other)
    return TRUE;
  if (info->finfo == NULL || other->finfo == NULL)
    return FALSE;
  if (GST_AUDIO_INFO_FORMAT (info) != GST_AUDIO_INFO_FORMAT (other))
    return FALSE;
  if (GST_AUDIO_INFO_FLAGS (info) != GST_AUDIO_INFO_FLAGS (other))
    return FALSE;
  if (GST_AUDIO_INFO_LAYOUT (info) != GST_AUDIO_INFO_LAYOUT (other))
    return FALSE;
  if (GST_AUDIO_INFO_RATE (info) != GST_AUDIO_INFO_RATE (other))
    return FALSE;
  if (GST_AUDIO_INFO_CHANNELS (info) != GST_AUDIO_INFO_CHANNELS (other))
    return FALSE;
  if (GST_AUDIO_INFO_CHANNELS (info) > 64)
    return TRUE;
  if (memcmp (info->position, other->position,
          GST_AUDIO_INFO_CHANNELS (info) * sizeof (GstAudioChannelPosition)) !=
      0)
    return FALSE;

  return TRUE;
}

开发者ID:reynaldo-samsung，项目名称:gst-plugins-base，代码行数:38，代码来源:audio-info.c

示例2: gst_pulse_fill_format_info

gboolean
gst_pulse_fill_format_info (GstAudioRingBufferSpec * spec, pa_format_info ** f,
    guint * channels)
{
  pa_format_info *format;
  pa_sample_format_t sf = PA_SAMPLE_INVALID;
  GstAudioInfo *ainfo = &spec->info;

  format = pa_format_info_new ();

  if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW
      && GST_AUDIO_INFO_WIDTH (ainfo) == 8) {
    format->encoding = PA_ENCODING_PCM;
    sf = PA_SAMPLE_ULAW;
  } else if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW
      && GST_AUDIO_INFO_WIDTH (ainfo) == 8) {
    format->encoding = PA_ENCODING_PCM;
    sf = PA_SAMPLE_ALAW;
  } else if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW) {
    format->encoding = PA_ENCODING_PCM;
    if (!gstaudioformat_to_pasampleformat (GST_AUDIO_INFO_FORMAT (ainfo), &sf))
      goto fail;
  } else if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3) {
    format->encoding = PA_ENCODING_AC3_IEC61937;
  } else if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3) {
    format->encoding = PA_ENCODING_EAC3_IEC61937;
  } else if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS) {
    format->encoding = PA_ENCODING_DTS_IEC61937;
  } else if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG) {
    format->encoding = PA_ENCODING_MPEG_IEC61937;
  } else {
    goto fail;
  }

  if (format->encoding == PA_ENCODING_PCM) {
    pa_format_info_set_sample_format (format, sf);
    pa_format_info_set_channels (format, GST_AUDIO_INFO_CHANNELS (ainfo));
  }

  pa_format_info_set_rate (format, GST_AUDIO_INFO_RATE (ainfo));

  if (!pa_format_info_valid (format))
    goto fail;

  *f = format;
  *channels = GST_AUDIO_INFO_CHANNELS (ainfo);

  return TRUE;

fail:
  if (format)
    pa_format_info_free (format);
  return FALSE;
}

开发者ID:lubing521，项目名称:gst-embedded-builder，代码行数:54，代码来源:pulseutil.c

示例3: gst_openal_src_prepare

static gboolean
gst_openal_src_prepare (GstAudioSrc * audiosrc, GstAudioRingBufferSpec * spec)
{
  GstOpenalSrc *openalsrc = GST_OPENAL_SRC (audiosrc);

  gst_openal_src_parse_spec (openalsrc, spec);
  if (openalsrc->format == AL_NONE) {
    GST_ELEMENT_ERROR (openalsrc, RESOURCE, SETTINGS, (NULL),
        ("Unable to get type %d, format %d, and %d channels", spec->type,
            GST_AUDIO_INFO_FORMAT (&spec->info),
            GST_AUDIO_INFO_CHANNELS (&spec->info)));
    return FALSE;
  }

  openalsrc->device =
      alcCaptureOpenDevice (openalsrc->default_device, openalsrc->rate,
      openalsrc->format, openalsrc->buffer_length);

  if (!openalsrc->device) {
    GST_ELEMENT_ERROR (openalsrc, RESOURCE, OPEN_READ,
        ("Could not open device."), GST_ALC_ERROR (openalsrc->device));
    return FALSE;
  }

  openalsrc->default_device_name =
      g_strdup (alcGetString (openalsrc->device, ALC_DEVICE_SPECIFIER));

  alcCaptureStart (openalsrc->device);

  return TRUE;
}

开发者ID:asrashley，项目名称:gst-plugins-bad，代码行数:31，代码来源:gstopenalsrc.c

示例4: gst_level_set_caps

static gboolean
gst_level_set_caps (GstBaseTransform * trans, GstCaps * in, GstCaps * out)
{
  GstLevel *filter = GST_LEVEL (trans);
  GstAudioInfo info;
  gint i, channels;

  if (!gst_audio_info_from_caps (&info, in))
    return FALSE;

  switch (GST_AUDIO_INFO_FORMAT (&info)) {
    case GST_AUDIO_FORMAT_S8:
      filter->process = gst_level_calculate_gint8;
      break;
    case GST_AUDIO_FORMAT_S16:
      filter->process = gst_level_calculate_gint16;
      break;
    case GST_AUDIO_FORMAT_S32:
      filter->process = gst_level_calculate_gint32;
      break;
    case GST_AUDIO_FORMAT_F32:
      filter->process = gst_level_calculate_gfloat;
      break;
    case GST_AUDIO_FORMAT_F64:
      filter->process = gst_level_calculate_gdouble;
      break;
    default:
      filter->process = NULL;
      break;
  }

  filter->info = info;

  channels = GST_AUDIO_INFO_CHANNELS (&info);

  /* allocate channel variable arrays */
  g_free (filter->CS);
  g_free (filter->peak);
  g_free (filter->last_peak);
  g_free (filter->decay_peak);
  g_free (filter->decay_peak_base);
  g_free (filter->decay_peak_age);
  filter->CS = g_new (gdouble, channels);
  filter->peak = g_new (gdouble, channels);
  filter->last_peak = g_new (gdouble, channels);
  filter->decay_peak = g_new (gdouble, channels);
  filter->decay_peak_base = g_new (gdouble, channels);

  filter->decay_peak_age = g_new (GstClockTime, channels);

  for (i = 0; i < channels; ++i) {
    filter->CS[i] = filter->peak[i] = filter->last_peak[i] =
        filter->decay_peak[i] = filter->decay_peak_base[i] = 0.0;
    filter->decay_peak_age[i] = G_GUINT64_CONSTANT (0);
  }

  gst_level_recalc_interval_frames (filter);

  return TRUE;
}

开发者ID:nnikos123，项目名称:gst-plugins-good，代码行数:60，代码来源:gstlevel.c

示例5: render_lines

static void
render_lines (GstAudioVisualizer * base, guint32 * vdata, gint16 * adata,
    guint num_samples)
{
  gint channels = GST_AUDIO_INFO_CHANNELS (&base->ainfo);
  guint i, c, s, x, y, oy;
  gfloat dx, dy;
  guint w = GST_VIDEO_INFO_WIDTH (&base->vinfo);
  guint h = GST_VIDEO_INFO_HEIGHT (&base->vinfo);
  gint x2, y2;

  /* draw lines */
  dx = (gfloat) (w - 1) / (gfloat) num_samples;
  dy = (h - 1) / 65536.0;
  oy = (h - 1) / 2;
  for (c = 0; c < channels; c++) {
    s = c;
    x2 = 0;
    y2 = (guint) (oy + (gfloat) adata[s] * dy);
    for (i = 1; i < num_samples; i++) {
      x = (guint) ((gfloat) i * dx);
      y = (guint) (oy + (gfloat) adata[s] * dy);
      s += channels;
      draw_line_aa (vdata, x2, x, y2, y, w, 0x00FFFFFF);
      x2 = x;
      y2 = y;
    }
  }
}

开发者ID:ego5710，项目名称:gst-plugins-bad，代码行数:29，代码来源:gstwavescope.c

示例6: gst_opus_enc_set_format

static gboolean
gst_opus_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
  GstOpusEnc *enc;

  enc = GST_OPUS_ENC (benc);

  g_mutex_lock (enc->property_lock);

  enc->n_channels = GST_AUDIO_INFO_CHANNELS (info);
  enc->sample_rate = GST_AUDIO_INFO_RATE (info);
  gst_opus_enc_setup_channel_mappings (enc, info);
  GST_DEBUG_OBJECT (benc, "Setup with %d channels, %d Hz", enc->n_channels,
      enc->sample_rate);

  /* handle reconfigure */
  if (enc->state) {
    opus_multistream_encoder_destroy (enc->state);
    enc->state = NULL;
  }
  if (!gst_opus_enc_setup (enc))
    return FALSE;

  enc->frame_samples = gst_opus_enc_get_frame_samples (enc);

  /* feedback to base class */
  gst_opus_enc_setup_base_class (enc, benc);

  g_mutex_unlock (enc->property_lock);

  return TRUE;
}

开发者ID:kanongil，项目名称:gst-plugins-bad，代码行数:32，代码来源:gstopusenc.c

示例7: gst_audio_filter_template_setup

static gboolean
gst_audio_filter_template_setup (GstAudioFilter * filter,
    const GstAudioInfo * info)
{
  GstAudioFilterTemplate *filter_template;
  GstAudioFormat fmt;
  gint chans, rate;

  filter_template = GST_AUDIO_FILTER_TEMPLATE (filter);

  rate = GST_AUDIO_INFO_RATE (info);
  chans = GST_AUDIO_INFO_CHANNELS (info);
  fmt = GST_AUDIO_INFO_FORMAT (info);

  GST_INFO_OBJECT (filter_template, "format %d (%s), rate %d, %d channels",
      fmt, GST_AUDIO_INFO_NAME (info), rate, chans);

  /* if any setup needs to be done (like memory allocated), do it here */

  /* The audio filter base class also saves the audio info in
   * GST_AUDIO_FILTER_INFO(filter) so it's automatically available
   * later from there as well */

  return TRUE;
}

开发者ID:johlim，项目名称:study，代码行数:25，代码来源:gstaudiofilter.c

示例8: gst_audio_panorama_set_caps

static gboolean
gst_audio_panorama_set_caps (GstBaseTransform * base, GstCaps * incaps,
    GstCaps * outcaps)
{
  GstAudioPanorama *filter = GST_AUDIO_PANORAMA (base);
  GstAudioInfo info;

  /*GST_INFO ("incaps are %" GST_PTR_FORMAT, incaps); */
  if (!gst_audio_info_from_caps (&info, incaps))
    goto no_format;

  GST_DEBUG ("try to process %d input with %d channels",
      GST_AUDIO_INFO_FORMAT (&info), GST_AUDIO_INFO_CHANNELS (&info));

  if (!gst_audio_panorama_set_process_function (filter, &info))
    goto no_format;

  filter->info = info;

  return TRUE;

no_format:
  {
    GST_DEBUG ("invalid caps");
    return FALSE;
  }
}

开发者ID:lubing521，项目名称:gst-embedded-builder，代码行数:27，代码来源:audiopanorama.c

示例9: gst_freeverb_set_caps

static gboolean
gst_freeverb_set_caps (GstBaseTransform * base, GstCaps * incaps,
                       GstCaps * outcaps)
{
    GstFreeverb *filter = GST_FREEVERB (base);
    GstAudioInfo info;

    /*GST_INFO ("incaps are %" GST_PTR_FORMAT, incaps); */
    if (!gst_audio_info_from_caps (&info, incaps))
        goto no_format;

    GST_DEBUG ("try to process %d input with %d channels",
               GST_AUDIO_INFO_FORMAT (&info), GST_AUDIO_INFO_CHANNELS (&info));

    if (!gst_freeverb_set_process_function (filter, &info))
        goto no_format;

    filter->info = info;

    gst_freeverb_init_rev_model (filter);
    filter->drained = FALSE;
    GST_INFO_OBJECT (base, "model configured");

    return TRUE;

no_format:
    {
        GST_DEBUG ("invalid caps");
        return FALSE;
    }
}

开发者ID:GrokImageCompression，项目名称:gst-plugins-bad，代码行数:31，代码来源:gstfreeverb.c

示例10: gst_audio_fx_base_fir_filter_setup

/* get notified of caps and plug in the correct process function */
static gboolean
gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base,
    const GstAudioInfo * info)
{
  GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);

  g_mutex_lock (&self->lock);
  if (self->buffer) {
    gst_audio_fx_base_fir_filter_push_residue (self);
    g_free (self->buffer);
    self->buffer = NULL;
    self->buffer_fill = 0;
    self->buffer_length = 0;
    self->start_ts = GST_CLOCK_TIME_NONE;
    self->start_off = GST_BUFFER_OFFSET_NONE;
    self->nsamples_out = 0;
    self->nsamples_in = 0;
  }

  gst_audio_fx_base_fir_filter_select_process_function (self,
      GST_AUDIO_INFO_FORMAT (info), GST_AUDIO_INFO_CHANNELS (info));
  g_mutex_unlock (&self->lock);

  return (self->process != NULL);
}

开发者ID:felipemogollon，项目名称:gst-plugins-good，代码行数:26，代码来源:audiofxbasefirfilter.c

示例11: render_color_lines

static void
render_color_lines (GstAudioVisualizer * base, guint32 * vdata,
    gint16 * adata, guint num_samples)
{
  GstWaveScope *scope = (GstWaveScope *) base;
  gint channels = GST_AUDIO_INFO_CHANNELS (&base->ainfo);
  guint i, c, s, x, y, oy;
  gfloat dx, dy;
  guint w = GST_VIDEO_INFO_WIDTH (&base->vinfo);
  guint h = GST_VIDEO_INFO_HEIGHT (&base->vinfo), h1 = h - 2;
  gdouble *flt = scope->flt;
  gint x2, y2, y3, y4;

  /* draw lines */
  dx = (gfloat) (w - 1) / (gfloat) num_samples;
  dy = (h - 1) / 65536.0;
  oy = (h - 1) / 2;
  for (c = 0; c < channels; c++) {
    s = c;

    /* do first pixels */
    x2 = 0;
    filter ((gfloat) adata[s]);

    y = (guint) (oy + flt[0] * dy);
    y2 = MIN (y, h1);

    y = (guint) (oy + flt[3] * dy);
    y3 = MIN (y, h1);

    y = (guint) (oy + (flt[4] + flt[5]) * dy);
    y4 = MIN (y, h1);

    for (i = 1; i < num_samples; i++) {
      x = (guint) ((gfloat) i * dx);
      filter ((gfloat) adata[s]);

      y = (guint) (oy + flt[0] * dy);
      y = MIN (y, h1);
      draw_line_aa (vdata, x2, x, y2, y, w, 0x00FF0000);
      y2 = y;

      y = (guint) (oy + flt[3] * dy);
      y = MIN (y, h1);
      draw_line_aa (vdata, x2, x, y3, y, w, 0x0000FF00);
      y3 = y;

      y = (guint) (oy + (flt[4] + flt[5]) * dy);
      y = MIN (y, h1);
      draw_line_aa (vdata, x2, x, y4, y, w, 0x000000FF);
      y4 = y;

      x2 = x;
      s += channels;
    }
    flt += 6;
  }
}

开发者ID:ego5710，项目名称:gst-plugins-bad，代码行数:58，代码来源:gstwavescope.c

示例12: gst_wave_scope_setup

static gboolean
gst_wave_scope_setup (GstAudioVisualizer * bscope)
{
  GstWaveScope *scope = GST_WAVE_SCOPE (bscope);

  if (scope->flt)
    g_free (scope->flt);

  scope->flt = g_new0 (gdouble, 6 * GST_AUDIO_INFO_CHANNELS (&bscope->ainfo));

  return TRUE;
}

开发者ID:ego5710，项目名称:gst-plugins-bad，代码行数:12，代码来源:gstwavescope.c

示例13: gst_audio_fx_base_iir_filter_setup

static gboolean
gst_audio_fx_base_iir_filter_setup (GstAudioFilter * base,
    const GstAudioInfo * info)
{
  GstAudioFXBaseIIRFilter *filter = GST_AUDIO_FX_BASE_IIR_FILTER (base);
  gboolean ret = TRUE;
  gint channels;

  g_mutex_lock (&filter->lock);
  switch (GST_AUDIO_INFO_FORMAT (info)) {
    case GST_AUDIO_FORMAT_F32:
      filter->process = (GstAudioFXBaseIIRFilterProcessFunc)
          process_32;
      break;
    case GST_AUDIO_FORMAT_F64:
      filter->process = (GstAudioFXBaseIIRFilterProcessFunc)
          process_64;
      break;
    default:
      ret = FALSE;
      break;
  }

  channels = GST_AUDIO_INFO_CHANNELS (info);

  if (channels != filter->nchannels) {
    guint i;
    GstAudioFXBaseIIRFilterChannelCtx *ctx;

    if (filter->channels) {
      for (i = 0; i < filter->nchannels; i++) {
        ctx = &filter->channels[i];

        g_free (ctx->x);
        g_free (ctx->y);
      }
      g_free (filter->channels);
    }

    filter->channels = g_new0 (GstAudioFXBaseIIRFilterChannelCtx, channels);
    for (i = 0; i < channels; i++) {
      ctx = &filter->channels[i];

      ctx->x = g_new0 (gdouble, filter->nb);
      ctx->y = g_new0 (gdouble, filter->na);
    }
    filter->nchannels = channels;
  }
  g_mutex_unlock (&filter->lock);

  return ret;
}

开发者ID:BigBrother-International，项目名称:gst-plugins-good，代码行数:52，代码来源:audiofxbaseiirfilter.c

示例14: gst_celt_enc_set_format

static gboolean
gst_celt_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
  GstCeltEnc *enc;
  GstCaps *otherpadcaps;

  enc = GST_CELT_ENC (benc);

  enc->channels = GST_AUDIO_INFO_CHANNELS (info);
  enc->rate = GST_AUDIO_INFO_RATE (info);

  /* handle reconfigure */
  if (enc->state) {
    celt_encoder_destroy (enc->state);
    enc->state = NULL;
  }
  if (enc->mode) {
    celt_mode_destroy (enc->mode);
    enc->mode = NULL;
  }
  memset (&enc->header, 0, sizeof (enc->header));

  otherpadcaps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (enc));
  if (otherpadcaps) {
    if (!gst_caps_is_empty (otherpadcaps)) {
      GstStructure *ps = gst_caps_get_structure (otherpadcaps, 0);
      gst_structure_get_int (ps, "frame-size", &enc->frame_size);
    }
    gst_caps_unref (otherpadcaps);
  }

  if (enc->requested_frame_size > 0)
    enc->frame_size = enc->requested_frame_size;

  GST_DEBUG_OBJECT (enc, "channels=%d rate=%d frame-size=%d",
      enc->channels, enc->rate, enc->frame_size);

  if (!gst_celt_enc_setup (enc))
    return FALSE;

  /* feedback to base class */
  gst_audio_encoder_set_latency (benc,
      gst_celt_enc_get_latency (enc), gst_celt_enc_get_latency (enc));
  gst_audio_encoder_set_frame_samples_min (benc, enc->frame_size);
  gst_audio_encoder_set_frame_samples_max (benc, enc->frame_size);
  gst_audio_encoder_set_frame_max (benc, 1);

  return TRUE;
}

开发者ID:dylansong77，项目名称:gstreamer，代码行数:49，代码来源:gstceltenc.c

示例15: gst_space_scope_render

static gboolean
gst_space_scope_render (GstAudioVisualizer * base, GstBuffer * audio,
                        GstVideoFrame * video)
{
    GstSpaceScope *scope = GST_SPACE_SCOPE (base);
    GstMapInfo amap;
    guint num_samples;

    gst_buffer_map (audio, &amap, GST_MAP_READ);

    num_samples =
        amap.size / (GST_AUDIO_INFO_CHANNELS (&base->ainfo) * sizeof (gint16));
    scope->process (base, (guint32 *) GST_VIDEO_FRAME_PLANE_DATA (video, 0),
                    (gint16 *) amap.data, num_samples);
    gst_buffer_unmap (audio, &amap);
    return TRUE;
}

开发者ID:reynaldo-samsung，项目名称:gst-plugins-bad，代码行数:17，代码来源:gstspacescope.c

示例16: pcm_config_from_spec

static void
pcm_config_from_spec (struct pcm_config *config,
    const GstAudioRingBufferSpec * spec)
{
  gint64 frames;

  config->format = pcm_format_from_gst (GST_AUDIO_INFO_FORMAT (&spec->info));
  config->channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
  config->rate = GST_AUDIO_INFO_RATE (&spec->info);

  gst_audio_info_convert (&spec->info,
      GST_FORMAT_TIME, spec->latency_time * GST_USECOND,
      GST_FORMAT_DEFAULT /* frames */ , &frames);

  config->period_size = frames;
  config->period_count = spec->buffer_time / spec->latency_time;
}

开发者ID:jhgorse，项目名称:gst-plugins-bad，代码行数:17，代码来源:tinyalsasink.c

示例17: gst_deinterleave_add_new_pads

static void
gst_deinterleave_add_new_pads (GstDeinterleave * self, GstCaps * caps)
{
  GstPad *pad;
  guint i;

  for (i = 0; i < GST_AUDIO_INFO_CHANNELS (&self->audio_info); i++) {
    gchar *name = g_strdup_printf ("src_%u", i);
    GstCaps *srccaps;
    GstAudioInfo info;
    GstAudioFormat format = GST_AUDIO_INFO_FORMAT (&self->audio_info);
    gint rate = GST_AUDIO_INFO_RATE (&self->audio_info);
    GstAudioChannelPosition position = GST_AUDIO_CHANNEL_POSITION_MONO;
    CopyStickyEventsData data;

    /* Set channel position if we know it */
    if (self->keep_positions)
      position = GST_AUDIO_INFO_POSITION (&self->audio_info, i);

    gst_audio_info_init (&info);
    gst_audio_info_set_format (&info, format, rate, 1, &position);

    srccaps = gst_audio_info_to_caps (&info);

    pad = gst_pad_new_from_static_template (&src_template, name);
    g_free (name);

    gst_pad_use_fixed_caps (pad);
    gst_pad_set_query_function (pad,
        GST_DEBUG_FUNCPTR (gst_deinterleave_src_query));
    gst_pad_set_active (pad, TRUE);

    data.pad = pad;
    data.caps = srccaps;
    gst_pad_sticky_events_foreach (self->sink, copy_sticky_events, &data);
    if (data.caps)
      gst_pad_set_caps (pad, data.caps);
    gst_element_add_pad (GST_ELEMENT (self), pad);
    self->srcpads = g_list_prepend (self->srcpads, gst_object_ref (pad));

    gst_caps_unref (srccaps);
  }

  gst_element_no_more_pads (GST_ELEMENT (self));
  self->srcpads = g_list_reverse (self->srcpads);
}

开发者ID:PeterXu，项目名称:gst-mobile，代码行数:46，代码来源:deinterleave.c

示例18: gst_chromaprint_transform_ip

static GstFlowReturn
gst_chromaprint_transform_ip (GstBaseTransform * trans, GstBuffer * buf)
{
  GstChromaprint *chromaprint = GST_CHROMAPRINT (trans);
  GstAudioFilter *filter = GST_AUDIO_FILTER (trans);
  GstMapInfo map_info;
  guint nsamples;
  gint rate, channels;

  rate = GST_AUDIO_INFO_RATE (&filter->info);
  channels = GST_AUDIO_INFO_CHANNELS (&filter->info);

  if (G_UNLIKELY (rate <= 0 || channels <= 0))
    return GST_FLOW_NOT_NEGOTIATED;

  if (!chromaprint->record)
    return GST_FLOW_OK;

  if (!gst_buffer_map (buf, &map_info, GST_MAP_READ))
    return GST_FLOW_ERROR;

  nsamples = map_info.size / (channels * 2);

  if (nsamples == 0)
    goto end;

  if (chromaprint->nsamples == 0) {
    chromaprint_start (chromaprint->context, rate, channels);
  }
  chromaprint->nsamples += nsamples;
  chromaprint->duration = chromaprint->nsamples / rate;

  chromaprint_feed (chromaprint->context, map_info.data,
      map_info.size / sizeof (guint16));

  if (chromaprint->duration >= chromaprint->max_duration
      && !chromaprint->fingerprint) {
    gst_chromaprint_create_fingerprint (chromaprint);
  }

end:
  gst_buffer_unmap (buf, &map_info);

  return GST_FLOW_OK;
}

开发者ID:asrashley，项目名称:gst-plugins-bad，代码行数:45，代码来源:gstchromaprint.c

示例19: gst_freeverb_set_process_function

static gboolean
gst_freeverb_set_process_function (GstFreeverb * filter, GstAudioInfo * info)
{
    gint channel_index, format_index;
    const GstAudioFormatInfo *finfo = info->finfo;

    /* set processing function */
    channel_index = GST_AUDIO_INFO_CHANNELS (info) - 1;
    if (channel_index > 1 || channel_index < 0) {
        filter->process = NULL;
        return FALSE;
    }

    format_index = GST_AUDIO_FORMAT_INFO_IS_FLOAT (finfo) ? 1 : 0;

    filter->process = process_functions[channel_index][format_index];
    return TRUE;
}

开发者ID:GrokImageCompression，项目名称:gst-plugins-bad，代码行数:18，代码来源:gstfreeverb.c

示例20: gst_deinterleave_chain

static GstFlowReturn
gst_deinterleave_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
  GstDeinterleave *self = GST_DEINTERLEAVE (parent);
  GstFlowReturn ret;

  g_return_val_if_fail (self->func != NULL, GST_FLOW_NOT_NEGOTIATED);
  g_return_val_if_fail (GST_AUDIO_INFO_WIDTH (&self->audio_info) > 0,
      GST_FLOW_NOT_NEGOTIATED);
  g_return_val_if_fail (GST_AUDIO_INFO_CHANNELS (&self->audio_info) > 0,
      GST_FLOW_NOT_NEGOTIATED);

  ret = gst_deinterleave_process (self, buffer);

  if (ret != GST_FLOW_OK)
    GST_DEBUG_OBJECT (self, "flow return: %s", gst_flow_get_name (ret));

  return ret;
}

开发者ID:PeterXu，项目名称:gst-mobile，代码行数:19，代码来源:deinterleave.c

注：本文中的GST_AUDIO_INFO_CHANNELS函数示例由纯净天空整理自Github/MSDocs等源码及文档管理平台，相关代码片段筛选自各路编程大神贡献的开源项目，源码版权归原作者所有，传播和使用请参考对应项目的License；未经允许，请勿转载。

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C++ GST_AUDIO_INFO_RATE函数代码示例发布时间：2022-05-30

C++ GST_AUDIO_DECODER函数代码示例发布时间：2022-05-30

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C++ GST_AUDIO_INFO_CHANNELS函数代码示例

示例1: gst_audio_info_is_equal

示例2: gst_pulse_fill_format_info

示例3: gst_openal_src_prepare

示例4: gst_level_set_caps

示例5: render_lines

示例6: gst_opus_enc_set_format

示例7: gst_audio_filter_template_setup

示例8: gst_audio_panorama_set_caps

示例9: gst_freeverb_set_caps

示例10: gst_audio_fx_base_fir_filter_setup

示例11: render_color_lines

示例12: gst_wave_scope_setup

示例13: gst_audio_fx_base_iir_filter_setup

示例14: gst_celt_enc_set_format

示例15: gst_space_scope_render

示例16: pcm_config_from_spec

示例17: gst_deinterleave_add_new_pads

示例18: gst_chromaprint_transform_ip

示例19: gst_freeverb_set_process_function

示例20: gst_deinterleave_chain

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