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audio: update for base class rename

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This commit is contained in:
Wim Taymans 2011-11-11 11:53:45 +01:00
parent 9daea802fa
commit 86e33bc46b
8 changed files with 43 additions and 43 deletions
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12
ext/jack/gstjackaudiosink.c
View file

@ -21,7 +21,7 @@
/**
* SECTION:element-jackaudiosink
* @see_also: #GstBaseAudioSink, #GstAudioRingBuffer
* @see_also: #GstAudioBaseSink, #GstAudioRingBuffer
*
* A Sink that outputs data to Jack ports.
*
@ -660,7 +660,7 @@ enum
};
#define gst_jack_audio_sink_parent_class parent_class
G_DEFINE_TYPE (GstJackAudioSink, gst_jack_audio_sink, GST_TYPE_BASE_AUDIO_SINK);
G_DEFINE_TYPE (GstJackAudioSink, gst_jack_audio_sink, GST_TYPE_AUDIO_BASE_SINK);
static void gst_jack_audio_sink_dispose (GObject * object);
static void gst_jack_audio_sink_set_property (GObject * object, guint prop_id,
@ -671,7 +671,7 @@ static void gst_jack_audio_sink_get_property (GObject * object, guint prop_id,
static GstCaps *gst_jack_audio_sink_getcaps (GstBaseSink * bsink,
GstCaps * filter);
static GstAudioRingBuffer
* gst_jack_audio_sink_create_ringbuffer (GstBaseAudioSink * sink);
* gst_jack_audio_sink_create_ringbuffer (GstAudioBaseSink * sink);
static void
gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass)
@ -679,7 +679,7 @@ gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass)
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSinkClass *gstbasesink_class;
GstBaseAudioSinkClass *gstbaseaudiosink_class;
GstAudioBaseSinkClass *gstbaseaudiosink_class;
GST_DEBUG_CATEGORY_INIT (gst_jack_audio_sink_debug, "jacksink", 0,
"jacksink element");
@ -687,7 +687,7 @@ gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass)
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesink_class = (GstBaseSinkClass *) klass;
gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;
gstbaseaudiosink_class = (GstAudioBaseSinkClass *) klass;
gobject_class->dispose = gst_jack_audio_sink_dispose;
gobject_class->get_property = gst_jack_audio_sink_get_property;
@ -857,7 +857,7 @@ no_client:
}
static GstAudioRingBuffer *
gst_jack_audio_sink_create_ringbuffer (GstBaseAudioSink * sink)
gst_jack_audio_sink_create_ringbuffer (GstAudioBaseSink * sink)
{
GstAudioRingBuffer *buffer;

4
ext/jack/gstjackaudiosink.h
View file

@ -48,7 +48,7 @@ typedef struct _GstJackAudioSinkClass GstJackAudioSinkClass;
* Opaque #GstJackAudioSink.
*/
struct _GstJackAudioSink {
GstBaseAudioSink element;
GstAudioBaseSink element;
/*< private >*/
/* cached caps */
@ -69,7 +69,7 @@ struct _GstJackAudioSink {
};
struct _GstJackAudioSinkClass {
GstBaseAudioSinkClass parent_class;
GstAudioBaseSinkClass parent_class;
};
GType gst_jack_audio_sink_get_type (void);

12
ext/jack/gstjackaudiosrc.c
View file

@ -42,7 +42,7 @@
/**
* SECTION:element-jackaudiosrc
* @see_also: #GstBaseAudioSrc, #GstAudioRingBuffer
* @see_also: #GstAudioBaseSrc, #GstAudioRingBuffer
*
* A Src that inputs data from Jack ports.
*
@ -678,7 +678,7 @@ static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
);
#define gst_jack_audio_src_parent_class parent_class
G_DEFINE_TYPE (GstJackAudioSrc, gst_jack_audio_src, GST_TYPE_BASE_AUDIO_SRC);
G_DEFINE_TYPE (GstJackAudioSrc, gst_jack_audio_src, GST_TYPE_AUDIO_BASE_SRC);
static void gst_jack_audio_src_dispose (GObject * object);
static void gst_jack_audio_src_set_property (GObject * object, guint prop_id,
@ -688,7 +688,7 @@ static void gst_jack_audio_src_get_property (GObject * object, guint prop_id,
static GstCaps *gst_jack_audio_src_getcaps (GstBaseSrc * bsrc,
GstCaps * filter);
static GstAudioRingBuffer *gst_jack_audio_src_create_ringbuffer (GstBaseAudioSrc
static GstAudioRingBuffer *gst_jack_audio_src_create_ringbuffer (GstAudioBaseSrc
* src);
/* GObject vmethod implementations */
@ -700,7 +700,7 @@ gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass)
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSrcClass *gstbasesrc_class;
GstBaseAudioSrcClass *gstbaseaudiosrc_class;
GstAudioBaseSrcClass *gstbaseaudiosrc_class;
GST_DEBUG_CATEGORY_INIT (gst_jack_audio_src_debug, "jacksrc", 0,
"jacksrc element");
@ -708,7 +708,7 @@ gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass)
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesrc_class = (GstBaseSrcClass *) klass;
gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
gstbaseaudiosrc_class = (GstAudioBaseSrcClass *) klass;
gobject_class->dispose = gst_jack_audio_src_dispose;
gobject_class->set_property = gst_jack_audio_src_set_property;
@ -880,7 +880,7 @@ no_client:
}
static GstAudioRingBuffer *
gst_jack_audio_src_create_ringbuffer (GstBaseAudioSrc * src)
gst_jack_audio_src_create_ringbuffer (GstAudioBaseSrc * src)
{
GstAudioRingBuffer *buffer;

4
ext/jack/gstjackaudiosrc.h
View file

@ -65,7 +65,7 @@ typedef struct _GstJackAudioSrcClass GstJackAudioSrcClass;
struct _GstJackAudioSrc
{
GstBaseAudioSrc src;
GstAudioBaseSrc src;
/*< private >*/
/* cached caps */
@ -87,7 +87,7 @@ struct _GstJackAudioSrc
struct _GstJackAudioSrcClass
{
GstBaseAudioSrcClass parent_class;
GstAudioBaseSrcClass parent_class;
};
GType gst_jack_audio_src_get_type (void);

2
ext/pulse/pulseaudiosink.c
View file

@ -818,7 +818,7 @@ gst_pulse_audio_sink_sink_acceptcaps (GstPulseAudioSink * pbin, GstPad * pad,
if (!gst_caps_is_fixed (caps))
goto out;
spec.latency_time = GST_BASE_AUDIO_SINK (pbin->psink)->latency_time;
spec.latency_time = GST_AUDIO_BASE_SINK (pbin->psink)->latency_time;
if (!gst_audio_ring_buffer_parse_caps (&spec, caps))
goto out;

44
ext/pulse/pulsesink.c
View file

@ -943,7 +943,7 @@ gst_pulseringbuffer_acquire (GstAudioRingBuffer * buf,
goto connect_failed;
/* our clock will now start from 0 again */
clock = GST_AUDIO_CLOCK (GST_BASE_AUDIO_SINK (psink)->provided_clock);
clock = GST_AUDIO_CLOCK (GST_AUDIO_BASE_SINK (psink)->provided_clock);
gst_audio_clock_reset (clock, 0);
if (!gst_pulsering_wait_for_stream_ready (psink, pbuf->stream))
@ -1173,7 +1173,7 @@ gst_pulseringbuffer_start (GstAudioRingBuffer * buf)
/* EOS needs running clock */
if (GST_BASE_SINK_CAST (psink)->eos ||
g_atomic_int_get (&GST_BASE_AUDIO_SINK (psink)->eos_rendering))
g_atomic_int_get (&GST_AUDIO_BASE_SINK (psink)->eos_rendering))
gst_pulsering_set_corked (pbuf, FALSE, FALSE);
pa_threaded_mainloop_unlock (mainloop);
@ -1751,7 +1751,7 @@ static GstStaticPadTemplate pad_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_IMPLEMENT_PULSEPROBE_METHODS (GstPulseSink, gst_pulsesink);
#define gst_pulsesink_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstPulseSink, gst_pulsesink, GST_TYPE_BASE_AUDIO_SINK,
G_DEFINE_TYPE_WITH_CODE (GstPulseSink, gst_pulsesink, GST_TYPE_AUDIO_BASE_SINK,
gst_pulsesink_init_contexts ();
G_IMPLEMENT_INTERFACE (GST_TYPE_PROPERTY_PROBE,
gst_pulsesink_property_probe_interface_init);
@ -1759,7 +1759,7 @@ G_DEFINE_TYPE_WITH_CODE (GstPulseSink, gst_pulsesink, GST_TYPE_BASE_AUDIO_SINK,
);
static GstAudioRingBuffer *
gst_pulsesink_create_ringbuffer (GstBaseAudioSink * sink)
gst_pulsesink_create_ringbuffer (GstAudioBaseSink * sink)
{
GstAudioRingBuffer *buffer;
@ -1771,7 +1771,7 @@ gst_pulsesink_create_ringbuffer (GstBaseAudioSink * sink)
}
static GstBuffer *
gst_pulsesink_payload (GstBaseAudioSink * sink, GstBuffer * buf)
gst_pulsesink_payload (GstAudioBaseSink * sink, GstBuffer * buf)
{
switch (sink->ringbuffer->spec.type) {
case GST_BUFTYPE_AC3:
@ -1820,7 +1820,7 @@ gst_pulsesink_class_init (GstPulseSinkClass * klass)
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstBaseSinkClass *gstbasesink_class = GST_BASE_SINK_CLASS (klass);
GstBaseSinkClass *bc;
GstBaseAudioSinkClass *gstaudiosink_class = GST_BASE_AUDIO_SINK_CLASS (klass);
GstAudioBaseSinkClass *gstaudiosink_class = GST_AUDIO_BASE_SINK_CLASS (klass);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
gobject_class->finalize = gst_pulsesink_finalize;
@ -1916,7 +1916,7 @@ gst_pulsesink_class_init (GstPulseSinkClass * klass)
/* returns the current time of the sink ringbuffer */
static GstClockTime
gst_pulsesink_get_time (GstClock * clock, GstBaseAudioSink * sink)
gst_pulsesink_get_time (GstClock * clock, GstAudioBaseSink * sink)
{
GstPulseSink *psink;
GstPulseRingBuffer *pbuf;
@ -2010,7 +2010,7 @@ done:
static gboolean
gst_pulsesink_query_acceptcaps (GstPulseSink * psink, GstCaps * caps)
{
GstPulseRingBuffer *pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK
GstPulseRingBuffer *pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK
(psink)->ringbuffer);
GstPad *pad = GST_BASE_SINK_PAD (psink);
GstCaps *pad_caps;
@ -2042,7 +2042,7 @@ gst_pulsesink_query_acceptcaps (GstPulseSink * psink, GstCaps * caps)
pa_threaded_mainloop_lock (mainloop);
spec.latency_time = GST_BASE_AUDIO_SINK (psink)->latency_time;
spec.latency_time = GST_AUDIO_BASE_SINK (psink)->latency_time;
if (!gst_audio_ring_buffer_parse_caps (&spec, caps))
goto out;
@ -2168,10 +2168,10 @@ gst_pulsesink_init (GstPulseSink * pulsesink)
pulsesink->proplist = NULL;
/* override with a custom clock */
if (GST_BASE_AUDIO_SINK (pulsesink)->provided_clock)
gst_object_unref (GST_BASE_AUDIO_SINK (pulsesink)->provided_clock);
if (GST_AUDIO_BASE_SINK (pulsesink)->provided_clock)
gst_object_unref (GST_AUDIO_BASE_SINK (pulsesink)->provided_clock);
GST_BASE_AUDIO_SINK (pulsesink)->provided_clock =
GST_AUDIO_BASE_SINK (pulsesink)->provided_clock =
gst_audio_clock_new ("GstPulseSinkClock",
(GstAudioClockGetTimeFunc) gst_pulsesink_get_time, pulsesink, NULL);
@ -2230,7 +2230,7 @@ gst_pulsesink_set_volume (GstPulseSink * psink, gdouble volume)
GST_DEBUG_OBJECT (psink, "setting volume to %f", volume);
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
if (pbuf == NULL || pbuf->stream == NULL)
goto no_buffer;
@ -2307,7 +2307,7 @@ gst_pulsesink_set_mute (GstPulseSink * psink, gboolean mute)
GST_DEBUG_OBJECT (psink, "setting mute state to %d", mute);
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
if (pbuf == NULL || pbuf->stream == NULL)
goto no_buffer;
@ -2400,7 +2400,7 @@ gst_pulsesink_get_volume (GstPulseSink * psink)
pa_threaded_mainloop_lock (mainloop);
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
if (pbuf == NULL || pbuf->stream == NULL)
goto no_buffer;
@ -2472,7 +2472,7 @@ gst_pulsesink_get_mute (GstPulseSink * psink)
pa_threaded_mainloop_lock (mainloop);
mute = psink->mute;
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
if (pbuf == NULL || pbuf->stream == NULL)
goto no_buffer;
@ -2534,7 +2534,7 @@ gst_pulsesink_device_description (GstPulseSink * psink)
goto no_mainloop;
pa_threaded_mainloop_lock (mainloop);
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
if (pbuf == NULL)
goto no_buffer;
@ -2667,7 +2667,7 @@ gst_pulsesink_change_title (GstPulseSink * psink, const gchar * t)
pa_threaded_mainloop_lock (mainloop);
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
if (pbuf == NULL || pbuf->stream == NULL)
goto no_buffer;
@ -2742,7 +2742,7 @@ gst_pulsesink_change_props (GstPulseSink * psink, GstTagList * l)
goto finish;
pa_threaded_mainloop_lock (mainloop);
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
if (pbuf == NULL || pbuf->stream == NULL)
goto no_buffer;
@ -2787,7 +2787,7 @@ gst_pulsesink_flush_ringbuffer (GstPulseSink * psink)
pa_threaded_mainloop_lock (mainloop);
pbuf = GST_PULSERING_BUFFER_CAST (GST_BASE_AUDIO_SINK (psink)->ringbuffer);
pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
if (pbuf == NULL || pbuf->stream == NULL)
goto no_buffer;
@ -2945,7 +2945,7 @@ gst_pulsesink_change_state (GstElement * element, GstStateChange transition)
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_element_post_message (element,
gst_message_new_clock_provide (GST_OBJECT_CAST (element),
GST_BASE_AUDIO_SINK (pulsesink)->provided_clock, TRUE));
GST_AUDIO_BASE_SINK (pulsesink)->provided_clock, TRUE));
break;
default:
@ -2961,7 +2961,7 @@ gst_pulsesink_change_state (GstElement * element, GstStateChange transition)
/* format_lost is reset in release() in baseaudiosink */
gst_element_post_message (element,
gst_message_new_clock_lost (GST_OBJECT_CAST (element),
GST_BASE_AUDIO_SINK (pulsesink)->provided_clock));
GST_AUDIO_BASE_SINK (pulsesink)->provided_clock));
break;
case GST_STATE_CHANGE_READY_TO_NULL:
gst_pulsesink_release_mainloop (pulsesink);

4
ext/pulse/pulsesink.h
View file

@ -56,7 +56,7 @@ typedef struct _GstPulseSinkClass GstPulseSinkClass;
struct _GstPulseSink
{
GstBaseAudioSink sink;
GstAudioBaseSink sink;
gchar *server, *device, *stream_name, *client_name;
gchar *device_description;
@ -87,7 +87,7 @@ struct _GstPulseSink
struct _GstPulseSinkClass
{
GstBaseAudioSinkClass parent_class;
GstAudioBaseSinkClass parent_class;
};
GType gst_pulsesink_get_type (void);

4
ext/pulse/pulsesrc.c
View file

@ -262,8 +262,8 @@ gst_pulsesrc_init (GstPulseSrc * pulsesrc)
pulsesrc->probe = gst_pulseprobe_new (G_OBJECT (pulsesrc), G_OBJECT_GET_CLASS (pulsesrc), PROP_DEVICE, pulsesrc->server, FALSE, TRUE); /* FALSE for sinks, TRUE for sources */
/* this should be the default but it isn't yet */
gst_base_audio_src_set_slave_method (GST_BASE_AUDIO_SRC (pulsesrc),
GST_BASE_AUDIO_SRC_SLAVE_SKEW);
gst_audio_base_src_set_slave_method (GST_AUDIO_BASE_SRC (pulsesrc),
GST_AUDIO_BASE_SRC_SLAVE_SKEW);
}
static void