/**
* SECTION:element-jackaudiosink
- * @see_also: #GstBaseAudioSink, #GstRingBuffer
+ * @see_also: #GstBaseAudioSink, #GstAudioRingBuffer
*
* A Sink that outputs data to Jack ports.
*
(GInstanceInitFunc) gst_jack_ring_buffer_init,
NULL
};
- GType tmp = g_type_register_static (GST_TYPE_RING_BUFFER,
+ GType tmp = g_type_register_static (GST_TYPE_AUDIO_RING_BUFFER,
"GstJackAudioSinkRingBuffer", &ringbuffer_info, 0);
g_once_init_leave (&ringbuffer_type, tmp);
}
static void
gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass)
{
- GstRingBufferClass *gstringbuffer_class;
+ GstAudioRingBufferClass *gstringbuffer_class;
- gstringbuffer_class = (GstRingBufferClass *) klass;
+ gstringbuffer_class = (GstAudioRingBufferClass *) klass;
ring_parent_class = g_type_class_peek_parent (klass);
jack_process_cb (jack_nframes_t nframes, void *arg)
{
GstJackAudioSink *sink;
- GstRingBuffer *buf;
+ GstAudioRingBuffer *buf;
gint readseg, len;
guint8 *readptr;
gint i, j, flen, channels;
sample_t *data;
- buf = GST_RING_BUFFER_CAST (arg);
+ buf = GST_AUDIO_RING_BUFFER_CAST (arg);
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
channels = GST_AUDIO_INFO_CHANNELS (&buf->spec.info);
(sample_t *) jack_port_get_buffer (sink->ports[i], nframes);
}
- if (gst_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) {
+ if (gst_audio_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) {
flen = len / channels;
/* the number of samples must be exactly the segment size */
}
/* clear written samples in the ringbuffer */
- gst_ring_buffer_clear (buf, readseg);
+ gst_audio_ring_buffer_clear (buf, readseg);
/* we wrote one segment */
- gst_ring_buffer_advance (buf, 1);
+ gst_audio_ring_buffer_advance (buf, 1);
} else {
GST_DEBUG_OBJECT (sink, "write %d frames silence", nframes);
/* We are not allowed to read from the ringbuffer, write silence to all
/* the _open_device method should make a connection with the server
*/
static gboolean
-gst_jack_ring_buffer_open_device (GstRingBuffer * buf)
+gst_jack_ring_buffer_open_device (GstAudioRingBuffer * buf)
{
GstJackAudioSink *sink;
jack_status_t status = 0;
/* close the connection with the server
*/
static gboolean
-gst_jack_ring_buffer_close_device (GstRingBuffer * buf)
+gst_jack_ring_buffer_close_device (GstAudioRingBuffer * buf)
{
GstJackAudioSink *sink;
* received for some reason, we fail here.
*/
static gboolean
-gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
+gst_jack_ring_buffer_acquire (GstAudioRingBuffer * buf,
+ GstAudioRingBufferSpec * spec)
{
GstJackAudioSink *sink;
GstJackRingBuffer *abuf;
/* function is called with LOCK */
static gboolean
-gst_jack_ring_buffer_release (GstRingBuffer * buf)
+gst_jack_ring_buffer_release (GstAudioRingBuffer * buf)
{
GstJackAudioSink *sink;
GstJackRingBuffer *abuf;
}
static gboolean
-gst_jack_ring_buffer_start (GstRingBuffer * buf)
+gst_jack_ring_buffer_start (GstAudioRingBuffer * buf)
{
GstJackAudioSink *sink;
}
static gboolean
-gst_jack_ring_buffer_pause (GstRingBuffer * buf)
+gst_jack_ring_buffer_pause (GstAudioRingBuffer * buf)
{
GstJackAudioSink *sink;
}
static gboolean
-gst_jack_ring_buffer_stop (GstRingBuffer * buf)
+gst_jack_ring_buffer_stop (GstAudioRingBuffer * buf)
{
GstJackAudioSink *sink;
#if defined (HAVE_JACK_0_120_1) || defined(HAVE_JACK_1_9_7)
static guint
-gst_jack_ring_buffer_delay (GstRingBuffer * buf)
+gst_jack_ring_buffer_delay (GstAudioRingBuffer * buf)
{
GstJackAudioSink *sink;
guint i, res = 0;
}
#else /* !(defined (HAVE_JACK_0_120_1) || defined(HAVE_JACK_1_9_7)) */
static guint
-gst_jack_ring_buffer_delay (GstRingBuffer * buf)
+gst_jack_ring_buffer_delay (GstAudioRingBuffer * buf)
{
GstJackAudioSink *sink;
guint i, res = 0;
static GstCaps *gst_jack_audio_sink_getcaps (GstBaseSink * bsink,
GstCaps * filter);
-static GstRingBuffer *gst_jack_audio_sink_create_ringbuffer (GstBaseAudioSink *
- sink);
+static GstAudioRingBuffer
+ * gst_jack_audio_sink_create_ringbuffer (GstBaseAudioSink * sink);
static void
gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass)
}
}
-static GstRingBuffer *
+static GstAudioRingBuffer *
gst_jack_audio_sink_create_ringbuffer (GstBaseAudioSink * sink)
{
- GstRingBuffer *buffer;
+ GstAudioRingBuffer *buffer;
buffer = g_object_new (GST_TYPE_JACK_RING_BUFFER, NULL);
GST_DEBUG_OBJECT (sink, "created ringbuffer @%p", buffer);