mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-18 07:47:17 +00:00
b45a1df5de
This are (almost) completely autogenerated from the documentation.
109 lines
2.7 KiB
C
109 lines
2.7 KiB
C
/* vim: set filetype=c: */
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% ClassName
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GstAudioSrc
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% TYPE_CLASS_NAME
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GST_TYPE_AUDIO_SRC
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% pads
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srcpad-audio
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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/gstaudiosrc.h>
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% prototypes
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static gboolean gst_replace_open (GstAudioSrc * src);
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static gboolean gst_replace_prepare (GstAudioSrc * src,
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GstAudioRingBufferSpec * spec);
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static gboolean gst_replace_unprepare (GstAudioSrc * src);
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static gboolean gst_replace_close (GstAudioSrc * src);
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static guint gst_replace_read (GstAudioSrc * src, gpointer data, guint length,
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GstClockTime * timestamp);
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static guint gst_replace_delay (GstAudioSrc * src);
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static void gst_replace_reset (GstAudioSrc * src);
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% declare-class
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GstAudioSrcClass *audio_src_class = GST_AUDIO_SRC_CLASS (klass);
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% set-methods
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audio_src_class->open = GST_DEBUG_FUNCPTR (gst_replace_open);
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audio_src_class->prepare = GST_DEBUG_FUNCPTR (gst_replace_prepare);
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audio_src_class->unprepare = GST_DEBUG_FUNCPTR (gst_replace_unprepare);
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audio_src_class->close = GST_DEBUG_FUNCPTR (gst_replace_close);
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audio_src_class->read = GST_DEBUG_FUNCPTR (gst_replace_read);
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audio_src_class->delay = GST_DEBUG_FUNCPTR (gst_replace_delay);
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audio_src_class->reset = GST_DEBUG_FUNCPTR (gst_replace_reset);
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% methods
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/* open the device with given specs */
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static gboolean
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gst_replace_open (GstAudioSrc * src)
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{
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GstReplace *replace = GST_REPLACE (src);
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GST_DEBUG_OBJECT (replace, "open");
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return TRUE;
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}
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/* prepare resources and state to operate with the given specs */
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static gboolean
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gst_replace_prepare (GstAudioSrc * src, GstAudioRingBufferSpec * spec)
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{
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GstReplace *replace = GST_REPLACE (src);
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GST_DEBUG_OBJECT (replace, "prepare");
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return TRUE;
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}
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/* undo anything that was done in prepare() */
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static gboolean
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gst_replace_unprepare (GstAudioSrc * src)
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{
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GstReplace *replace = GST_REPLACE (src);
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GST_DEBUG_OBJECT (replace, "unprepare");
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return TRUE;
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}
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/* close the device */
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static gboolean
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gst_replace_close (GstAudioSrc * src)
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{
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GstReplace *replace = GST_REPLACE (src);
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GST_DEBUG_OBJECT (replace, "close");
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return TRUE;
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}
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/* read samples from the device */
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static guint
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gst_replace_read (GstAudioSrc * src, gpointer data, guint length,
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GstClockTime * timestamp)
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{
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GstReplace *replace = GST_REPLACE (src);
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GST_DEBUG_OBJECT (replace, "read");
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return 0;
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}
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/* get number of samples queued in the device */
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static guint
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gst_replace_delay (GstAudioSrc * src)
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{
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GstReplace *replace = GST_REPLACE (src);
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GST_DEBUG_OBJECT (replace, "delay");
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return 0;
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}
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/* reset the audio device, unblock from a write */
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static void
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gst_replace_reset (GstAudioSrc * src)
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{
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GstReplace *replace = GST_REPLACE (src);
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GST_DEBUG_OBJECT (replace, "reset");
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}
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% end
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