Commit e7c3ddf3 authored by Stefan Kost's avatar Stefan Kost

gst-libs/gst/audio/gstbaseaudiosink.c

Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_change_state):
Fix typo in comment.
* gst/playback/gstdecodebin.c (gst_decode_bin_class_init,
free_dynamics, pad_probe, close_pad_link, try_to_link_1,
get_our_ghost_pad, remove_element_chain, queue_underrun_cb,
close_link):
* gst/playback/gstplaybin.c (gst_play_bin_set_property,
gen_audio_element, remove_sinks, gst_play_bin_send_event_to_sink):
Remove trailing whitespaces in comments.
* gst/volume/Makefile.am:
Fix tabs.
parent 16b8bd4c
2007-05-18 Stefan Kost <ensonic@users.sf.net>
* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_change_state):
Fix typo in comment.
* gst/playback/gstdecodebin.c (gst_decode_bin_class_init,
free_dynamics, pad_probe, close_pad_link, try_to_link_1,
get_our_ghost_pad, remove_element_chain, queue_underrun_cb,
close_link):
* gst/playback/gstplaybin.c (gst_play_bin_set_property,
gen_audio_element, remove_sinks, gst_play_bin_send_event_to_sink):
Remove trailing whitespaces in comments.
* gst/volume/Makefile.am:
Fix tabs.
2007-05-18 Stefan Kost <ensonic@users.sf.net>
patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
......@@ -1321,7 +1321,7 @@ gst_base_audio_sink_change_state (GstElement * element,
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* slop slaving ourselves to the master, if any */
/* stop slaving ourselves to the master, if any */
gst_clock_set_master (sink->provided_clock, NULL);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
......
......@@ -237,7 +237,7 @@ gst_decode_bin_class_init (GstDecodeBinClass * klass)
/* check if the bin is dynamic.
*
* If there are no outstanding dynamic connections, the bin is
* If there are no outstanding dynamic connections, the bin is
* considered to be non-dynamic.
*/
static gboolean
......@@ -496,7 +496,7 @@ free_dynamics (GstDecodeBin * decode_bin)
}
/* this function runs through the element factories and returns a list
* of all elements that are able to sink the given caps
* of all elements that are able to sink the given caps
*/
static GList *
find_compatibles (GstDecodeBin * decode_bin, const GstCaps * caps)
......@@ -670,7 +670,7 @@ pad_probe (GstPad * pad, GstMiniObject * data, GstDecodeBin * decode_bin)
* If the pad has a raw format, this function will create a ghostpad
* for the pad onto the decodebin.
*
* If no compatible elements could be found, this function will signal
* If no compatible elements could be found, this function will signal
* the unknown_type signal.
*/
static void
......@@ -798,7 +798,7 @@ setup_caps_delay:
}
}
/* Decide whether an element is a demuxer based on the
/* Decide whether an element is a demuxer based on the
* klass and number/type of src pad templates it has */
static gboolean
is_demuxer_element (GstElement * srcelement)
......@@ -911,7 +911,7 @@ try_to_link_1 (GstDecodeBin * decode_bin, GstElement * srcelement, GstPad * pad,
}
/* try to link the given pad to a sinkpad */
/* FIXME, find the sinkpad by looping over the pads instead of
/* FIXME, find the sinkpad by looping over the pads instead of
* looking it up by name */
if ((sinkpad = gst_element_get_pad (element, "sink")) == NULL) {
/* if no pad is found we can't do anything */
......@@ -967,7 +967,7 @@ try_to_link_1 (GstDecodeBin * decode_bin, GstElement * srcelement, GstPad * pad,
/* get rid of the sinkpad now */
gst_object_unref (sinkpad);
/* Set the queue to paused and set the pointer to NULL so we don't
/* Set the queue to paused and set the pointer to NULL so we don't
* remove it below */
if (queue != NULL) {
gst_element_set_state (queue, GST_STATE_PAUSED);
......@@ -1047,7 +1047,7 @@ get_our_ghost_pad (GstDecodeBin * decode_bin, GstPad * pad)
}
/* remove all downstream elements starting from the given pad.
* Also make sure to remove the ghostpad we created for the raw
* Also make sure to remove the ghostpad we created for the raw
* decoded stream.
*/
static void
......@@ -1069,7 +1069,7 @@ remove_element_chain (GstDecodeBin * decode_bin, GstPad * pad)
GST_DEBUG_OBJECT (decode_bin, "%s:%s", GST_DEBUG_PAD_NAME (pad));
int_links = gst_pad_get_internal_links (pad);
/* remove all elements linked to this pad up to the ghostpad
/* remove all elements linked to this pad up to the ghostpad
* that we created for this stream */
for (walk = int_links; walk; walk = g_list_next (walk)) {
GstPad *pad;
......@@ -1133,7 +1133,7 @@ remove_element_chain (GstDecodeBin * decode_bin, GstPad * pad)
gst_bin_remove (GST_BIN (decode_bin), elem);
}
/* there are @bytes bytes in @queue, enlarge it
/* there are @bytes bytes in @queue, enlarge it
*
* Returns: new max number of bytes in @queue
*/
......@@ -1166,7 +1166,7 @@ queue_underrun_cb (GstElement * queue, GstDecodeBin * decode_bin)
{
/* FIXME: we don't really do anything here for now. Ideally we should
* see if some of the queues are filled and increase their values
* in that case.
* in that case.
* Note: be very carefull with thread safety here as this underrun
* signal is done from the streaming thread of queue srcpad which
* is different from the pad_added (where we add the queue to the
......@@ -1496,7 +1496,7 @@ close_link (GstElement * element, GstDecodeBin * decode_bin)
}
/* Check if this is an element with more than 1 pad. If this element
* has more than 1 pad, we need to be carefull not to signal the
* has more than 1 pad, we need to be carefull not to signal the
* no_more_pads signal after connecting the first pad. */
more = g_list_length (to_connect) > 1;
......
......@@ -26,7 +26,7 @@
* audio and/or video player.
* </para>
* <para>
* It can handle both audio and video files and features
* It can handle both audio and video files and features
* <itemizedlist>
* <listitem>
* automatic file type recognition and based on that automatic
......@@ -644,7 +644,7 @@ gst_play_bin_set_property (GObject * object, guint prop_id,
gst_object_sink (GST_OBJECT_CAST (play_bin->video_sink));
}
/* when changing the videosink, we just remove the
* video pipeline from the cache so that it will be
* video pipeline from the cache so that it will be
* regenerated with the new sink element */
g_hash_table_remove (play_bin->cache, "vbin");
break;
......@@ -1131,7 +1131,7 @@ link_failed:
}
/* make the element (bin) that contains the elements needed to perform
* visualisation ouput. The idea is to split the audio using tee, then
* visualisation ouput. The idea is to split the audio using tee, then
* sending the output to the regular audio bin and the other output to
* the vis plugin that transforms it into a video that is rendered with the
* normal video bin. The video and audio bins are run in threads to make sure
......@@ -1385,7 +1385,7 @@ remove_sinks (GstPlayBin * play_bin)
* media file. First we count the number of audio and video streams.
* If there is no video stream but there exists an audio stream,
* we install a visualisation pipeline.
*
*
* Also make sure to only connect the first audio and video pad. FIXME
* this should eventually be handled with a tuner interface so that
* one can switch the streams.
......@@ -1660,7 +1660,7 @@ gst_play_bin_send_event_to_sink (GstPlayBin * play_bin, GstEvent * event)
return res;
}
/* We only want to send the event to a single sink (overriding GstBin's
/* We only want to send the event to a single sink (overriding GstBin's
* behaviour), but we want to keep GstPipeline's behaviour - wrapping seek
* events appropriately. So, this is a messy duplication of code. */
static gboolean
......
......@@ -6,7 +6,7 @@ libgstvolume_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstvolume_la_LIBADD = \
$(top_builddir)/gst-libs/gst/interfaces/libgstinterfaces-$(GST_MAJORMINOR).la \
$(GST_BASE_LIBS) \
$(GST_CONTROLLER_LIBS) \
$(GST_CONTROLLER_LIBS) \
$(GST_LIBS) \
$(LIBOIL_LIBS)
......
Markdown is supported
0% or .
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment