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authorStefan Kost <ensonic@users.sf.net>2011-04-27 16:56:09 +0300
committerStefan Kost <ensonic@users.sf.net>2011-05-18 10:31:38 +0300
commitc46725845b91631b7b7bbd44765f592c3c83408b (patch)
tree8b53b8cd59b0f74c50aca865e4ec726adce41f3f /tools
parentf76feeb6321d0ec205049d3543f1e7949c294f99 (diff)
downloadgstreamer-plugins-bad-c46725845b91631b7b7bbd44765f592c3c83408b.tar.gz
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.
Diffstat (limited to 'tools')
-rw-r--r--tools/element-templates/audiofilter31
1 files changed, 30 insertions, 1 deletions
diff --git a/tools/element-templates/audiofilter b/tools/element-templates/audiofilter
index cf10fc120..5e7ea376b 100644
--- a/tools/element-templates/audiofilter
+++ b/tools/element-templates/audiofilter
@@ -12,27 +12,56 @@ gstreamer-audio-0.10
% prototypes
static gboolean
gst_replace_setup (GstAudioFilter * filter, GstRingBufferSpec * format);
+static gboolean
+gst_replace_start (GstBaseTransform * trans);
static GstFlowReturn
gst_replace_transform_ip (GstBaseTransform * trans, GstBuffer * buf);
+static gboolean
+gst_replace_stop (GstBaseTransform * trans);
% declare-class
GstAudioFilterClass *audio_filter_class = GST_AUDIO_FILTER_CLASS (klass);
GstBaseTransformClass *base_transform_class = GST_BASE_TRANSFORM_CLASS (klass);
% set-methods
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->stop = GST_DEBUG_FUNCPTR (gst_replace_stop);
% methods
static gboolean
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;
}
static GstFlowReturn
gst_replace_transform_ip (GstBaseTransform * trans, GstBuffer * buf)
{
+ /* GstReplace *replace = GST_REPLACE (trans); */
+
+ /* process the audio data in the buffer in-place */
+ return GST_FLOW_OK;
+}
- return GST_FLOW_ERROR;
+static gboolean
+gst_replace_stop (GstBaseTransform * trans)
+{
+ /* GstReplace *replace = GST_REPLACE (trans); */
+
+ /* finalize processing */
+ return TRUE;
}
% end