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update for audiobase* rename

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This commit is contained in:
Wim Taymans 2011-11-11 12:01:17 +01:00
parent 86e33bc46b
commit 3d9d2c6c05
4 changed files with 9 additions and 9 deletions
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6
ext/jack/gstjackaudiosink.c
View file

@ -679,7 +679,7 @@ gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass)
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSinkClass *gstbasesink_class;
GstAudioBaseSinkClass *gstbaseaudiosink_class;
GstAudioBaseSinkClass *gstaudiobasesink_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 = (GstAudioBaseSinkClass *) klass;
gstaudiobasesink_class = (GstAudioBaseSinkClass *) klass;
gobject_class->dispose = gst_jack_audio_sink_dispose;
gobject_class->get_property = gst_jack_audio_sink_get_property;
@ -719,7 +719,7 @@ gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass)
gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_sink_getcaps);
gstbaseaudiosink_class->create_ringbuffer =
gstaudiobasesink_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_jack_audio_sink_create_ringbuffer);
/* ref class from a thread-safe context to work around missing bit of

2
ext/jack/gstjackaudiosink.h
View file

@ -25,7 +25,7 @@
#include <jack/jack.h>
#include <gst/gst.h>
#include <gst/audio/gstbaseaudiosink.h>
#include <gst/audio/gstaudiobasesink.h>
#include "gstjack.h"
#include "gstjackaudioclient.h"

6
ext/jack/gstjackaudiosrc.c
View file

@ -700,7 +700,7 @@ gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass)
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSrcClass *gstbasesrc_class;
GstAudioBaseSrcClass *gstbaseaudiosrc_class;
GstAudioBaseSrcClass *gstaudiobasesrc_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 = (GstAudioBaseSrcClass *) klass;
gstaudiobasesrc_class = (GstAudioBaseSrcClass *) klass;
gobject_class->dispose = gst_jack_audio_src_dispose;
gobject_class->set_property = gst_jack_audio_src_set_property;
@ -739,7 +739,7 @@ gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass)
"Tristan Matthews <tristan@sat.qc.ca>");
gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_src_getcaps);
gstbaseaudiosrc_class->create_ringbuffer =
gstaudiobasesrc_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_jack_audio_src_create_ringbuffer);
/* ref class from a thread-safe context to work around missing bit of

4
ext/pulse/pulsesink.c
View file

@ -2958,7 +2958,7 @@ gst_pulsesink_change_state (GstElement * element, GstStateChange transition)
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
/* format_lost is reset in release() in baseaudiosink */
/* format_lost is reset in release() in audiobasesink */
gst_element_post_message (element,
gst_message_new_clock_lost (GST_OBJECT_CAST (element),
GST_AUDIO_BASE_SINK (pulsesink)->provided_clock));
@ -2983,7 +2983,7 @@ mainloop_failed:
state_failure:
{
if (transition == GST_STATE_CHANGE_NULL_TO_READY) {
/* Clear the PA mainloop if baseaudiosink failed to open the ring_buffer */
/* Clear the PA mainloop if audiobasesink failed to open the ring_buffer */
g_assert (mainloop);
gst_pulsesink_release_mainloop (pulsesink);
}