C++ (Cpp) GST_AUDIO_INVERT 예제들

예제 #1

파일: audioinvert.c 프로젝트: ConfusedReality/pkg_multimedia_gst-plugins-good

/* GstBaseTransform vmethod implementations */
static GstFlowReturn
gst_audio_invert_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
  GstAudioInvert *filter = GST_AUDIO_INVERT (base);
  guint num_samples;
  GstClockTime timestamp, stream_time;
  GstMapInfo map;

  timestamp = GST_BUFFER_TIMESTAMP (buf);
  stream_time =
      gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);

  GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
      GST_TIME_ARGS (timestamp));

  if (GST_CLOCK_TIME_IS_VALID (stream_time))
    gst_object_sync_values (GST_OBJECT (filter), stream_time);

  if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
    return GST_FLOW_OK;

  gst_buffer_map (buf, &map, GST_MAP_READWRITE);
  num_samples = map.size / GST_AUDIO_FILTER_BPS (filter);

  filter->process (filter, map.data, num_samples);

  gst_buffer_unmap (buf, &map);

  return GST_FLOW_OK;
}

예제 #2

파일: audioinvert.c 프로젝트: dgerlach/gst-plugins-good

/* GstBaseTransform vmethod implementations */
static GstFlowReturn
gst_audio_invert_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
  GstAudioInvert *filter = GST_AUDIO_INVERT (base);
  guint num_samples;
  GstClockTime timestamp, stream_time;

  timestamp = GST_BUFFER_TIMESTAMP (buf);
  stream_time =
      gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);

  GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
      GST_TIME_ARGS (timestamp));

  if (GST_CLOCK_TIME_IS_VALID (stream_time))
    gst_object_sync_values (G_OBJECT (filter), stream_time);

  num_samples =
      GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);

  if (gst_base_transform_is_passthrough (base) ||
      G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
    return GST_FLOW_OK;

  filter->process (filter, GST_BUFFER_DATA (buf), num_samples);

  return GST_FLOW_OK;
}

예제 #3

파일: audioinvert.c 프로젝트: ConfusedReality/pkg_multimedia_gst-plugins-good

static void
gst_audio_invert_get_property (GObject * object, guint prop_id,
    GValue * value, GParamSpec * pspec)
{
  GstAudioInvert *filter = GST_AUDIO_INVERT (object);

  switch (prop_id) {
    case PROP_DEGREE:
      g_value_set_float (value, filter->degree);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

예제 #4

파일: audioinvert.c 프로젝트: ConfusedReality/pkg_multimedia_gst-plugins-good

static void
gst_audio_invert_set_property (GObject * object, guint prop_id,
    const GValue * value, GParamSpec * pspec)
{
  GstAudioInvert *filter = GST_AUDIO_INVERT (object);

  switch (prop_id) {
    case PROP_DEGREE:
      filter->degree = g_value_get_float (value);
      gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (filter),
          filter->degree == 0.0);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

예제 #5

파일: audioinvert.c 프로젝트: dgerlach/gst-plugins-good

static gboolean
gst_audio_invert_setup (GstAudioFilter * base, GstRingBufferSpec * format)
{
  GstAudioInvert *filter = GST_AUDIO_INVERT (base);
  gboolean ret = TRUE;

  if (format->type == GST_BUFTYPE_FLOAT && format->width == 32)
    filter->process = (GstAudioInvertProcessFunc)
        gst_audio_invert_transform_float;
  else if (format->type == GST_BUFTYPE_LINEAR && format->width == 16)
    filter->process = (GstAudioInvertProcessFunc)
        gst_audio_invert_transform_int;
  else
    ret = FALSE;

  return ret;
}

예제 #6

파일: audioinvert.c 프로젝트: ConfusedReality/pkg_multimedia_gst-plugins-good

static gboolean
gst_audio_invert_setup (GstAudioFilter * base, const GstAudioInfo * info)
{
  GstAudioInvert *filter = GST_AUDIO_INVERT (base);
  gboolean ret = TRUE;

  switch (GST_AUDIO_INFO_FORMAT (info)) {
    case GST_AUDIO_FORMAT_S16:
      filter->process = (GstAudioInvertProcessFunc)
          gst_audio_invert_transform_int;
      break;
    case GST_AUDIO_FORMAT_F32:
      filter->process = (GstAudioInvertProcessFunc)
          gst_audio_invert_transform_float;
      break;
    default:
      ret = FALSE;
      break;
  }
  return ret;
}