element-templates: improve the audiofilter template

Add comments. Add start/stop methods. Add (commented) instance casts at the
begin of the method. Make transform_ip returning FLOW_OK by default.
This commit is contained in:
Stefan Kost 2011-04-27 16:56:09 +03:00
parent f76feeb632
commit c46725845b

View file

@ -12,27 +12,56 @@ gstreamer-audio-0.10
% prototypes % prototypes
static gboolean static gboolean
gst_replace_setup (GstAudioFilter * filter, GstRingBufferSpec * format); gst_replace_setup (GstAudioFilter * filter, GstRingBufferSpec * format);
static gboolean
gst_replace_start (GstBaseTransform * trans);
static GstFlowReturn static GstFlowReturn
gst_replace_transform_ip (GstBaseTransform * trans, GstBuffer * buf); gst_replace_transform_ip (GstBaseTransform * trans, GstBuffer * buf);
static gboolean
gst_replace_stop (GstBaseTransform * trans);
% declare-class % declare-class
GstAudioFilterClass *audio_filter_class = GST_AUDIO_FILTER_CLASS (klass); GstAudioFilterClass *audio_filter_class = GST_AUDIO_FILTER_CLASS (klass);
GstBaseTransformClass *base_transform_class = GST_BASE_TRANSFORM_CLASS (klass); GstBaseTransformClass *base_transform_class = GST_BASE_TRANSFORM_CLASS (klass);
% set-methods % set-methods
audio_filter_class->setup = GST_DEBUG_FUNCPTR (gst_replace_setup); audio_filter_class->setup = GST_DEBUG_FUNCPTR (gst_replace_setup);
base_transform_class->start = GST_DEBUG_FUNCPTR (gst_replace_start);
base_transform_class->transform_ip = GST_DEBUG_FUNCPTR (gst_replace_transform_ip); base_transform_class->transform_ip = GST_DEBUG_FUNCPTR (gst_replace_transform_ip);
base_transform_class->stop = GST_DEBUG_FUNCPTR (gst_replace_stop);
% methods % methods
static gboolean static gboolean
gst_replace_setup (GstAudioFilter * filter, GstRingBufferSpec * format) gst_replace_setup (GstAudioFilter * filter, GstRingBufferSpec * format)
{ {
/* GstReplace *replace = GST_REPLACE (filter); */
/* handle audio format changes */
return TRUE;
}
static gboolean
gst_replace_start (GstBaseTransform * trans)
{
/* GstReplace *replace = GST_REPLACE (trans); */
/* initialize processing */
return TRUE; return TRUE;
} }
static GstFlowReturn static GstFlowReturn
gst_replace_transform_ip (GstBaseTransform * trans, GstBuffer * buf) gst_replace_transform_ip (GstBaseTransform * trans, GstBuffer * buf)
{ {
/* GstReplace *replace = GST_REPLACE (trans); */
return GST_FLOW_ERROR; /* process the audio data in the buffer in-place */
return GST_FLOW_OK;
}
static gboolean
gst_replace_stop (GstBaseTransform * trans)
{
/* GstReplace *replace = GST_REPLACE (trans); */
/* finalize processing */
return TRUE;
} }
% end % end