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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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192 lines
5.2 KiB
Text
192 lines
5.2 KiB
Text
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/* vim: set filetype=c: */
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% ClassName
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GstAudioDecoder
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% TYPE_CLASS_NAME
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GST_TYPE_AUDIO_DECODER
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% pads
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srcpad-audio
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sinkpad-simple
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% pkg-config
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gstreamer-audio-1.0
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% includes
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#include <gst/audio/gstaudiodecoder.h>
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% prototypes
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static gboolean gst_replace_start (GstAudioDecoder * decoder);
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static gboolean gst_replace_stop (GstAudioDecoder * decoder);
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static gboolean gst_replace_set_format (GstAudioDecoder * decoder, GstCaps * caps);
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static GstFlowReturn gst_replace_parse (GstAudioDecoder * decoder,
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GstAdapter * adapter, gint * offset, gint * length);
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static GstFlowReturn gst_replace_handle_frame (GstAudioDecoder * decoder,
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GstBuffer * buffer);
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static void gst_replace_flush (GstAudioDecoder * decoder, gboolean hard);
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static GstFlowReturn gst_replace_pre_push (GstAudioDecoder * decoder,
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GstBuffer ** buffer);
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static gboolean gst_replace_sink_event (GstAudioDecoder * decoder,
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GstEvent * event);
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static gboolean gst_replace_src_event (GstAudioDecoder * decoder, GstEvent * event);
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static gboolean gst_replace_open (GstAudioDecoder * decoder);
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static gboolean gst_replace_close (GstAudioDecoder * decoder);
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static gboolean gst_replace_negotiate (GstAudioDecoder * decoder);
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static gboolean gst_replace_decide_allocation (GstAudioDecoder * decoder,
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GstQuery * query);
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static gboolean gst_replace_propose_allocation (GstAudioDecoder * decoder,
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GstQuery * query);
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% declare-class
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GstAudioDecoderClass *audio_decoder_class = GST_AUDIO_DECODER_CLASS (klass);
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% set-methods
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audio_decoder_class->start = GST_DEBUG_FUNCPTR (gst_replace_start);
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audio_decoder_class->stop = GST_DEBUG_FUNCPTR (gst_replace_stop);
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audio_decoder_class->set_format = GST_DEBUG_FUNCPTR (gst_replace_set_format);
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audio_decoder_class->parse = GST_DEBUG_FUNCPTR (gst_replace_parse);
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audio_decoder_class->handle_frame = GST_DEBUG_FUNCPTR (gst_replace_handle_frame);
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audio_decoder_class->flush = GST_DEBUG_FUNCPTR (gst_replace_flush);
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audio_decoder_class->pre_push = GST_DEBUG_FUNCPTR (gst_replace_pre_push);
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audio_decoder_class->sink_event = GST_DEBUG_FUNCPTR (gst_replace_sink_event);
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audio_decoder_class->src_event = GST_DEBUG_FUNCPTR (gst_replace_src_event);
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audio_decoder_class->open = GST_DEBUG_FUNCPTR (gst_replace_open);
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audio_decoder_class->close = GST_DEBUG_FUNCPTR (gst_replace_close);
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audio_decoder_class->negotiate = GST_DEBUG_FUNCPTR (gst_replace_negotiate);
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audio_decoder_class->decide_allocation = GST_DEBUG_FUNCPTR (gst_replace_decide_allocation);
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audio_decoder_class->propose_allocation = GST_DEBUG_FUNCPTR (gst_replace_propose_allocation);
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% methods
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static gboolean
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gst_replace_start (GstAudioDecoder * decoder)
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{
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GstReplace *replace = GST_REPLACE (decoder);
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GST_DEBUG_OBJECT (replace, "start");
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return TRUE;
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}
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static gboolean
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gst_replace_stop (GstAudioDecoder * decoder)
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{
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GstReplace *replace = GST_REPLACE (decoder);
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GST_DEBUG_OBJECT (replace, "stop");
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return TRUE;
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}
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static gboolean
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gst_replace_set_format (GstAudioDecoder * decoder, GstCaps * caps)
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{
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GstReplace *replace = GST_REPLACE (decoder);
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GST_DEBUG_OBJECT (replace, "set_format");
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return TRUE;
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}
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static GstFlowReturn
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gst_replace_parse (GstAudioDecoder * decoder, GstAdapter * adapter, gint * offset,
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gint * length)
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{
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GstReplace *replace = GST_REPLACE (decoder);
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GST_DEBUG_OBJECT (replace, "parse");
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return GST_FLOW_OK;
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}
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static GstFlowReturn
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gst_replace_handle_frame (GstAudioDecoder * decoder, GstBuffer * buffer)
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{
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GstReplace *replace = GST_REPLACE (decoder);
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GST_DEBUG_OBJECT (replace, "handle_frame");
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return GST_FLOW_OK;
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}
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static void
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gst_replace_flush (GstAudioDecoder * decoder, gboolean hard)
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{
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GstReplace *replace = GST_REPLACE (decoder);
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GST_DEBUG_OBJECT (replace, "flush");
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}
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static GstFlowReturn
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gst_replace_pre_push (GstAudioDecoder * decoder, GstBuffer ** buffer)
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{
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GstReplace *replace = GST_REPLACE (decoder);
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GST_DEBUG_OBJECT (replace, "pre_push");
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return GST_FLOW_OK;
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}
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static gboolean
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gst_replace_sink_event (GstAudioDecoder * decoder, GstEvent * event)
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{
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GstReplace *replace = GST_REPLACE (decoder);
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GST_DEBUG_OBJECT (replace, "sink_event");
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return TRUE;
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}
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static gboolean
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gst_replace_src_event (GstAudioDecoder * decoder, GstEvent * event)
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{
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GstReplace *replace = GST_REPLACE (decoder);
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GST_DEBUG_OBJECT (replace, "src_event");
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return TRUE;
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}
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static gboolean
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gst_replace_open (GstAudioDecoder * decoder)
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{
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GstReplace *replace = GST_REPLACE (decoder);
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GST_DEBUG_OBJECT (replace, "open");
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return TRUE;
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}
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static gboolean
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gst_replace_close (GstAudioDecoder * decoder)
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{
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GstReplace *replace = GST_REPLACE (decoder);
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GST_DEBUG_OBJECT (replace, "close");
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return TRUE;
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}
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static gboolean
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gst_replace_negotiate (GstAudioDecoder * decoder)
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{
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GstReplace *replace = GST_REPLACE (decoder);
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GST_DEBUG_OBJECT (replace, "negotiate");
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return TRUE;
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}
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static gboolean
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gst_replace_decide_allocation (GstAudioDecoder * decoder, GstQuery * query)
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{
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GstReplace *replace = GST_REPLACE (decoder);
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GST_DEBUG_OBJECT (replace, "decide_allocation");
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return TRUE;
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}
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static gboolean
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gst_replace_propose_allocation (GstAudioDecoder * decoder, GstQuery * query)
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{
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GstReplace *replace = GST_REPLACE (decoder);
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GST_DEBUG_OBJECT (replace, "propose_allocation");
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return TRUE;
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}
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% end
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