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spectrum: port to 0.11

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* use G_DEFINE_TYPE
* adjust to new GstBuffer
This commit is contained in:
Mark Nauwelaerts 2011-06-30 10:51:55 +02:00
parent 696ffc85d7
commit 2d9bcbdf03
1 changed files with 23 additions and 25 deletions
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48
gst/spectrum/gstspectrum.c
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@ -162,8 +162,8 @@ enum
PROP_MULTI_CHANNEL
};
GST_BOILERPLATE (GstSpectrum, gst_spectrum, GstAudioFilter,
GST_TYPE_AUDIO_FILTER);
#define gst_spectrum_parent_class parent_class
G_DEFINE_TYPE (GstSpectrum, gst_spectrum, GST_TYPE_AUDIO_FILTER);
static void gst_spectrum_finalize (GObject * object);
static void gst_spectrum_set_property (GObject * object, guint prop_id,
@ -177,31 +177,14 @@ static GstFlowReturn gst_spectrum_transform_ip (GstBaseTransform * trans,
static gboolean gst_spectrum_setup (GstAudioFilter * base,
GstRingBufferSpec * format);
static void
gst_spectrum_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
GstCaps *caps;
gst_element_class_set_details_simple (element_class, "Spectrum analyzer",
"Filter/Analyzer/Audio",
"Run an FFT on the audio signal, output spectrum data",
"Erik Walthinsen <omega@cse.ogi.edu>, "
"Stefan Kost <ensonic@users.sf.net>, "
"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
caps = gst_caps_from_string (ALLOWED_CAPS);
gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (g_class),
caps);
gst_caps_unref (caps);
}
static void
gst_spectrum_class_init (GstSpectrumClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstBaseTransformClass *trans_class = GST_BASE_TRANSFORM_CLASS (klass);
GstAudioFilterClass *filter_class = GST_AUDIO_FILTER_CLASS (klass);
GstCaps *caps;
gobject_class->set_property = gst_spectrum_set_property;
gobject_class->get_property = gst_spectrum_get_property;
@ -277,10 +260,21 @@ gst_spectrum_class_init (GstSpectrumClass * klass)
GST_DEBUG_CATEGORY_INIT (gst_spectrum_debug, "spectrum", 0,
"audio spectrum analyser element");
gst_element_class_set_details_simple (element_class, "Spectrum analyzer",
"Filter/Analyzer/Audio",
"Run an FFT on the audio signal, output spectrum data",
"Erik Walthinsen <omega@cse.ogi.edu>, "
"Stefan Kost <ensonic@users.sf.net>, "
"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
caps = gst_caps_from_string (ALLOWED_CAPS);
gst_audio_filter_class_add_pad_templates (filter_class, caps);
gst_caps_unref (caps);
}
static void
gst_spectrum_init (GstSpectrum * spectrum, GstSpectrumClass * g_class)
gst_spectrum_init (GstSpectrum * spectrum)
{
spectrum->post_messages = DEFAULT_POST_MESSAGES;
spectrum->message_magnitude = DEFAULT_MESSAGE_MAGNITUDE;
@ -1011,15 +1005,17 @@ gst_spectrum_transform_ip (GstBaseTransform * trans, GstBuffer * buffer)
guint nfft = 2 * bands - 2;
guint input_pos;
gfloat *input;
const guint8 *data = GST_BUFFER_DATA (buffer);
guint size = GST_BUFFER_SIZE (buffer);
const guint8 *data, *mdata;
gsize size;
guint frame_size = width * channels;
guint fft_todo, msg_todo, block_size;
gboolean have_full_interval;
GstSpectrumChannel *cd;
GstSpectrumInputData input_data;
GST_LOG_OBJECT (spectrum, "input size: %d bytes", GST_BUFFER_SIZE (buffer));
data = mdata = gst_buffer_map (buffer, &size, NULL, GST_MAP_READ);
GST_LOG_OBJECT (spectrum, "input size: %" G_GSIZE_FORMAT " bytes", size);
if (GST_BUFFER_IS_DISCONT (buffer)) {
GST_DEBUG_OBJECT (spectrum, "Discontinuity detected -- flushing");
@ -1145,6 +1141,8 @@ gst_spectrum_transform_ip (GstBaseTransform * trans, GstBuffer * buffer)
spectrum->input_pos = input_pos;
gst_buffer_unmap (buffer, (guint8 *) mdata, -1);
g_assert (size == 0);
return GST_FLOW_OK;