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jack: port jack elements

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This commit is contained in:
Wim Taymans 2011-04-26 15:42:47 +02:00
parent 9a96783abb
commit 5ed18ad7b9
2 changed files with 38 additions and 47 deletions
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43
ext/jack/gstjackaudiosink.c
View file

@ -437,8 +437,8 @@ gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
buffer_size, spec->segsize, spec->segtotal); buffer_size, spec->segsize, spec->segtotal);
/* allocate the ringbuffer memory now */ /* allocate the ringbuffer memory now */
buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize); buf->size = spec->segtotal * spec->segsize;
memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data)); buf->memory = g_malloc0 (buf->size);
if ((res = gst_jack_audio_client_set_active (sink->client, TRUE))) if ((res = gst_jack_audio_client_set_active (sink->client, TRUE)))
goto could_not_activate; goto could_not_activate;
@ -540,8 +540,8 @@ gst_jack_ring_buffer_release (GstRingBuffer * buf)
abuf->sample_rate = -1; abuf->sample_rate = -1;
/* free the buffer */ /* free the buffer */
gst_buffer_unref (buf->data); g_free (buf->memory);
buf->data = NULL; buf->memory = NULL;
return TRUE; return TRUE;
} }
@ -655,11 +655,8 @@ enum
PROP_LAST PROP_LAST
}; };
#define _do_init(bla) \ #define gst_jack_audio_sink_parent_class parent_class
GST_DEBUG_CATEGORY_INIT (gst_jack_audio_sink_debug, "jacksink", 0, "jacksink element"); G_DEFINE_TYPE (GstJackAudioSink, gst_jack_audio_sink, GST_TYPE_BASE_AUDIO_SINK);
GST_BOILERPLATE_FULL (GstJackAudioSink, gst_jack_audio_sink, GstBaseAudioSink,
GST_TYPE_BASE_AUDIO_SINK, _do_init);
static void gst_jack_audio_sink_dispose (GObject * object); static void gst_jack_audio_sink_dispose (GObject * object);
static void gst_jack_audio_sink_set_property (GObject * object, guint prop_id, static void gst_jack_audio_sink_set_property (GObject * object, guint prop_id,
@ -671,27 +668,19 @@ static GstCaps *gst_jack_audio_sink_getcaps (GstBaseSink * bsink);
static GstRingBuffer *gst_jack_audio_sink_create_ringbuffer (GstBaseAudioSink * static GstRingBuffer *gst_jack_audio_sink_create_ringbuffer (GstBaseAudioSink *
sink); sink);
static void
gst_jack_audio_sink_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_details_simple (element_class, "Audio Sink (Jack)",
"Sink/Audio", "Output audio to a JACK server",
"Wim Taymans <wim.taymans@gmail.com>");
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&jackaudiosink_sink_factory));
}
static void static void
gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass) gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass)
{ {
GObjectClass *gobject_class; GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSinkClass *gstbasesink_class; GstBaseSinkClass *gstbasesink_class;
GstBaseAudioSinkClass *gstbaseaudiosink_class; GstBaseAudioSinkClass *gstbaseaudiosink_class;
GST_DEBUG_CATEGORY_INIT (gst_jack_audio_sink_debug, "jacksink", 0,
"jacksink element");
gobject_class = (GObjectClass *) klass; gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesink_class = (GstBaseSinkClass *) klass; gstbasesink_class = (GstBaseSinkClass *) klass;
gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass; gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;
@ -716,6 +705,13 @@ gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass)
GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE | GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
G_PARAM_STATIC_STRINGS)); G_PARAM_STATIC_STRINGS));
gst_element_class_set_details_simple (gstelement_class, "Audio Sink (Jack)",
"Sink/Audio", "Output audio to a JACK server",
"Wim Taymans <wim.taymans@gmail.com>");
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&jackaudiosink_sink_factory));
gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_sink_getcaps); gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_sink_getcaps);
gstbaseaudiosink_class->create_ringbuffer = gstbaseaudiosink_class->create_ringbuffer =
@ -729,8 +725,7 @@ gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass)
} }
static void static void
gst_jack_audio_sink_init (GstJackAudioSink * sink, gst_jack_audio_sink_init (GstJackAudioSink * sink)
GstJackAudioSinkClass * g_class)
{ {
sink->connect = DEFAULT_PROP_CONNECT; sink->connect = DEFAULT_PROP_CONNECT;
sink->server = g_strdup (DEFAULT_PROP_SERVER); sink->server = g_strdup (DEFAULT_PROP_SERVER);

42
ext/jack/gstjackaudiosrc.c
View file

@ -448,8 +448,8 @@ gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
buffer_size, spec->segsize, spec->segtotal); buffer_size, spec->segsize, spec->segtotal);
/* allocate the ringbuffer memory now */ /* allocate the ringbuffer memory now */
buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize); buf->size = spec->segtotal * spec->segsize;
memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data)); buf->memory = g_malloc0 (buf->size);
if ((res = gst_jack_audio_client_set_active (src->client, TRUE))) if ((res = gst_jack_audio_client_set_active (src->client, TRUE)))
goto could_not_activate; goto could_not_activate;
@ -553,8 +553,8 @@ gst_jack_ring_buffer_release (GstRingBuffer * buf)
abuf->sample_rate = -1; abuf->sample_rate = -1;
/* free the buffer */ /* free the buffer */
gst_buffer_unref (buf->data); g_free (buf->memory);
buf->data = NULL; buf->memory = NULL;
return TRUE; return TRUE;
} }
@ -674,11 +674,8 @@ static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]") "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
); );
#define _do_init(bla) \ #define gst_jack_audio_src_parent_class parent_class
GST_DEBUG_CATEGORY_INIT(gst_jack_audio_src_debug, "jacksrc", 0, "jacksrc element"); G_DEFINE_TYPE (GstJackAudioSrc, gst_jack_audio_src, GST_TYPE_BASE_AUDIO_SRC);
GST_BOILERPLATE_FULL (GstJackAudioSrc, gst_jack_audio_src, GstBaseAudioSrc,
GST_TYPE_BASE_AUDIO_SRC, _do_init);
static void gst_jack_audio_src_dispose (GObject * object); static void gst_jack_audio_src_dispose (GObject * object);
static void gst_jack_audio_src_set_property (GObject * object, guint prop_id, static void gst_jack_audio_src_set_property (GObject * object, guint prop_id,
@ -692,28 +689,20 @@ static GstRingBuffer *gst_jack_audio_src_create_ringbuffer (GstBaseAudioSrc *
/* GObject vmethod implementations */ /* GObject vmethod implementations */
static void
gst_jack_audio_src_base_init (gpointer gclass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (gclass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_set_details_simple (element_class, "Audio Source (Jack)",
"Source/Audio", "Captures audio from a JACK server",
"Tristan Matthews <tristan@sat.qc.ca>");
}
/* initialize the jack_audio_src's class */ /* initialize the jack_audio_src's class */
static void static void
gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass) gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass)
{ {
GObjectClass *gobject_class; GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSrcClass *gstbasesrc_class; GstBaseSrcClass *gstbasesrc_class;
GstBaseAudioSrcClass *gstbaseaudiosrc_class; GstBaseAudioSrcClass *gstbaseaudiosrc_class;
gobject_class = (GObjectClass *) klass; GST_DEBUG_CATEGORY_INIT (gst_jack_audio_src_debug, "jacksrc", 0,
"jacksrc element");
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesrc_class = (GstBaseSrcClass *) klass; gstbasesrc_class = (GstBaseSrcClass *) klass;
gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass; gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
@ -738,6 +727,13 @@ gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass)
GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE | GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
G_PARAM_STATIC_STRINGS)); G_PARAM_STATIC_STRINGS));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_set_details_simple (gstelement_class, "Audio Source (Jack)",
"Source/Audio", "Captures audio from a JACK server",
"Tristan Matthews <tristan@sat.qc.ca>");
gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_src_getcaps); gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_src_getcaps);
gstbaseaudiosrc_class->create_ringbuffer = gstbaseaudiosrc_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_jack_audio_src_create_ringbuffer); GST_DEBUG_FUNCPTR (gst_jack_audio_src_create_ringbuffer);
@ -755,7 +751,7 @@ gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass)
* initialize instance structure * initialize instance structure
*/ */
static void static void
gst_jack_audio_src_init (GstJackAudioSrc * src, GstJackAudioSrcClass * gclass) gst_jack_audio_src_init (GstJackAudioSrc * src)
{ {
//gst_base_src_set_live(GST_BASE_SRC (src), TRUE); //gst_base_src_set_live(GST_BASE_SRC (src), TRUE);
src->connect = DEFAULT_PROP_CONNECT; src->connect = DEFAULT_PROP_CONNECT;