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/* vim: set filetype=c: */
% ClassName
GstAudioEncoder
% TYPE_CLASS_NAME
GST_TYPE_AUDIO_ENCODER
% pads
srcpad-simple
sinkpad-audio
% pkg-config
gstreamer-audio-1.0
% includes
#include <gst/audio/gstaudioencoder.h>
% prototypes
static gboolean gst_replace_start (GstAudioEncoder * encoder);
static gboolean gst_replace_stop (GstAudioEncoder * encoder);
static gboolean gst_replace_set_format (GstAudioEncoder * encoder,
GstAudioInfo * info);
static GstFlowReturn gst_replace_handle_frame (GstAudioEncoder * encoder,
GstBuffer * buffer);
static void gst_replace_flush (GstAudioEncoder * encoder);
static GstFlowReturn gst_replace_pre_push (GstAudioEncoder * encoder,
GstBuffer ** buffer);
static gboolean gst_replace_sink_event (GstAudioEncoder * encoder,
GstEvent * event);
static gboolean gst_replace_src_event (GstAudioEncoder * encoder, GstEvent * event);
static GstCaps *gst_replace_getcaps (GstAudioEncoder * encoder, GstCaps * filter);
static gboolean gst_replace_open (GstAudioEncoder * encoder);
static gboolean gst_replace_close (GstAudioEncoder * encoder);
static gboolean gst_replace_negotiate (GstAudioEncoder * encoder);
static gboolean gst_replace_decide_allocation (GstAudioEncoder * encoder,
GstQuery * query);
static gboolean gst_replace_propose_allocation (GstAudioEncoder * encoder,
GstQuery * query);
% declare-class
GstAudioEncoderClass *audio_encoder_class = GST_AUDIO_ENCODER_CLASS (klass);
% set-methods
audio_encoder_class->start = GST_DEBUG_FUNCPTR (gst_replace_start);
audio_encoder_class->stop = GST_DEBUG_FUNCPTR (gst_replace_stop);
audio_encoder_class->set_format = GST_DEBUG_FUNCPTR (gst_replace_set_format);
audio_encoder_class->handle_frame = GST_DEBUG_FUNCPTR (gst_replace_handle_frame);
audio_encoder_class->flush = GST_DEBUG_FUNCPTR (gst_replace_flush);
audio_encoder_class->pre_push = GST_DEBUG_FUNCPTR (gst_replace_pre_push);
audio_encoder_class->sink_event = GST_DEBUG_FUNCPTR (gst_replace_sink_event);
audio_encoder_class->src_event = GST_DEBUG_FUNCPTR (gst_replace_src_event);
audio_encoder_class->getcaps = GST_DEBUG_FUNCPTR (gst_replace_getcaps);
audio_encoder_class->open = GST_DEBUG_FUNCPTR (gst_replace_open);
audio_encoder_class->close = GST_DEBUG_FUNCPTR (gst_replace_close);
audio_encoder_class->negotiate = GST_DEBUG_FUNCPTR (gst_replace_negotiate);
audio_encoder_class->decide_allocation = GST_DEBUG_FUNCPTR (gst_replace_decide_allocation);
audio_encoder_class->propose_allocation = GST_DEBUG_FUNCPTR (gst_replace_propose_allocation);
% methods
static gboolean
gst_replace_start (GstAudioEncoder * encoder)
{
GstReplace *replace = GST_REPLACE (encoder);
GST_DEBUG_OBJECT (replace, "start");
return TRUE;
}
static gboolean
gst_replace_stop (GstAudioEncoder * encoder)
{
GstReplace *replace = GST_REPLACE (encoder);
GST_DEBUG_OBJECT (replace, "stop");
return TRUE;
}
static gboolean
gst_replace_set_format (GstAudioEncoder * encoder, GstAudioInfo * info)
{
GstReplace *replace = GST_REPLACE (encoder);
GST_DEBUG_OBJECT (replace, "set_format");
return TRUE;
}
static GstFlowReturn
gst_replace_handle_frame (GstAudioEncoder * encoder, GstBuffer * buffer)
{
GstReplace *replace = GST_REPLACE (encoder);
GST_DEBUG_OBJECT (replace, "handle_frame");
return GST_FLOW_OK;
}
static void
gst_replace_flush (GstAudioEncoder * encoder)
{
GstReplace *replace = GST_REPLACE (encoder);
GST_DEBUG_OBJECT (replace, "flush");
}
static GstFlowReturn
gst_replace_pre_push (GstAudioEncoder * encoder, GstBuffer ** buffer)
{
GstReplace *replace = GST_REPLACE (encoder);
GST_DEBUG_OBJECT (replace, "pre_push");
return GST_FLOW_OK;
}
static gboolean
gst_replace_sink_event (GstAudioEncoder * encoder, GstEvent * event)
{
GstReplace *replace = GST_REPLACE (encoder);
GST_DEBUG_OBJECT (replace, "sink_event");
return TRUE;
}
static gboolean
gst_replace_src_event (GstAudioEncoder * encoder, GstEvent * event)
{
GstReplace *replace = GST_REPLACE (encoder);
GST_DEBUG_OBJECT (replace, "src_event");
return TRUE;
}
static GstCaps *
gst_replace_getcaps (GstAudioEncoder * encoder, GstCaps * filter)
{
GstReplace *replace = GST_REPLACE (encoder);
GST_DEBUG_OBJECT (replace, "getcaps");
return NULL;
}
static gboolean
gst_replace_open (GstAudioEncoder * encoder)
{
GstReplace *replace = GST_REPLACE (encoder);
GST_DEBUG_OBJECT (replace, "open");
return TRUE;
}
static gboolean
gst_replace_close (GstAudioEncoder * encoder)
{
GstReplace *replace = GST_REPLACE (encoder);
GST_DEBUG_OBJECT (replace, "close");
return TRUE;
}
static gboolean
gst_replace_negotiate (GstAudioEncoder * encoder)
{
GstReplace *replace = GST_REPLACE (encoder);
GST_DEBUG_OBJECT (replace, "negotiate");
return TRUE;
}
static gboolean
gst_replace_decide_allocation (GstAudioEncoder * encoder, GstQuery * query)
{
GstReplace *replace = GST_REPLACE (encoder);
GST_DEBUG_OBJECT (replace, "decide_allocation");
return TRUE;
}
static gboolean
gst_replace_propose_allocation (GstAudioEncoder * encoder, GstQuery * query)
{
GstReplace *replace = GST_REPLACE (encoder);
GST_DEBUG_OBJECT (replace, "propose_allocation");
return TRUE;
}
% end
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