audiobase*: Drop trailing withespaces

gst-libs/gst/audio/audio.c

@@ -71,7 +71,7 @@ gst_audio_buffer_clip (GstBuffer * buffer, GstSegment * segment, gint rate,
     /* No timestamp - assume the buffer is completely in the segment */
     return buffer;
 
-  /* Get copies of the buffer metadata to change later. 
+  /* Get copies of the buffer metadata to change later.
    * Calculate the missing values for the calculations,
    * they won't be changed later though. */
 

gst-libs/gst/audio/gstaudiobasesrc.c

@@ -2,7 +2,7 @@
  * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
  *                    2005 Wim Taymans <wim@fluendo.com>
  *
- * gstaudiobasesrc.c: 
+ * gstaudiobasesrc.c:
  *
  * This library is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Library General Public
@@ -247,7 +247,7 @@ gst_audio_base_src_init (GstAudioBaseSrc * audiobasesrc)
   else
     GST_OBJECT_FLAG_UNSET (audiobasesrc, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
   audiobasesrc->priv->slave_method = DEFAULT_SLAVE_METHOD;
-  /* reset blocksize we use latency time to calculate a more useful 
+  /* reset blocksize we use latency time to calculate a more useful
    * value based on negotiated format. */
   GST_BASE_SRC (audiobasesrc)->blocksize = 0;
 
@@ -359,7 +359,7 @@ gst_audio_base_src_get_time (GstClock * clock, GstAudioBaseSrc * src)
  * @src: a #GstAudioBaseSrc
  * @provide: new state
  *
- * Controls whether @src will provide a clock or not. If @provide is %TRUE, 
+ * Controls whether @src will provide a clock or not. If @provide is %TRUE,
  * gst_element_provide_clock() will return a clock that reflects the datarate
  * of @src. If @provide is %FALSE, gst_element_provide_clock() will return NULL.
  */

gst-libs/gst/audio/gstaudiocdsrc.c

@@ -1102,7 +1102,7 @@ gst_audio_cd_src_uri_handler_init (gpointer g_iface, gpointer iface_data)
  * gst_audio_cd_src_add_track:
  * @src: a #GstAudioCdSrc
  * @track: address of #GstAudioCdSrcTrack to add
- * 
+ *
  * CDDA sources use this function from their start vfunc to announce the
  * available data and audio tracks to the base source class. The caller
  * should allocate @track on the stack, the base source will do a shallow

gst-libs/gst/audio/gstaudiodecoder.h

@@ -268,7 +268,7 @@ struct _GstAudioDecoderClass
                                        GstEvent *event);
 
   gboolean      (*open)               (GstAudioDecoder *dec);
-  
+
   gboolean      (*close)              (GstAudioDecoder *dec);
 
   gboolean      (*negotiate)          (GstAudioDecoder *dec);

gst-libs/gst/audio/gstaudioencoder.c

@@ -89,7 +89,7 @@
  * </orderedlist>
  *
  * Subclass is responsible for providing pad template caps for
- * source and sink pads. The pads need to be named "sink" and "src". It also 
+ * source and sink pads. The pads need to be named "sink" and "src". It also
  * needs to set the fixed caps on srcpad, when the format is ensured.  This
  * is typically when base class calls subclass' @set_format function, though
  * it might be delayed until calling @gst_audio_encoder_finish_frame.
@@ -2640,7 +2640,7 @@ gst_audio_encoder_negotiate_unlocked (GstAudioEncoder * enc)
  * Negotiate with downstream elements to currently configured #GstCaps.
  * Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if
  * negotiate fails.
- * 
+ *
  * Returns: #TRUE if the negotiation succeeded, else #FALSE.
  */
 gboolean

gst-libs/gst/audio/gstaudioringbuffer.c

@@ -1214,7 +1214,7 @@ gst_audio_ring_buffer_set_sample (GstAudioRingBuffer * buf, guint64 sample)
   if (G_UNLIKELY (buf->samples_per_seg == 0))
     return;
 
-  /* FIXME, we assume the ringbuffer can restart at a random 
+  /* FIXME, we assume the ringbuffer can restart at a random
    * position, round down to the beginning and keep track of
    * offset when calculating the processed samples. */
   buf->segbase = buf->segdone - sample / buf->samples_per_seg;
@@ -1661,7 +1661,7 @@ gst_audio_ring_buffer_commit (GstAudioRingBuffer * buf, guint64 * sample,
  * @len: the number of samples in data to read
  * @timestamp: where the timestamp is returned
  *
- * Read @len samples from the ringbuffer into the memory pointed 
+ * Read @len samples from the ringbuffer into the memory pointed
  * to by @data.
  * The first sample should be read from position @sample in
  * the ringbuffer.
@@ -1850,7 +1850,7 @@ gst_audio_ring_buffer_prepare_read (GstAudioRingBuffer * buf, gint * segment,
  * @buf: the #GstAudioRingBuffer to advance
  * @advance: the number of segments written
  *
- * Subclasses should call this function to notify the fact that 
+ * Subclasses should call this function to notify the fact that
  * @advance segments are now processed by the device.
  *
  * MT safe.
@@ -1915,7 +1915,7 @@ gst_audio_ring_buffer_clear (GstAudioRingBuffer * buf, gint segment)
  * @allowed: the new value
  *
  * Tell the ringbuffer that it is allowed to start playback when
- * the ringbuffer is filled with samples. 
+ * the ringbuffer is filled with samples.
  *
  * MT safe.
  */

gst-libs/gst/audio/gstaudiosink.c

@@ -47,7 +47,7 @@
  *   </varlistentry>
  *   <varlistentry>
  *     <term>delay()</term>
- *     <listitem><para>Get the number of samples written but not yet played 
+ *     <listitem><para>Get the number of samples written but not yet played
  *     by the device.</para></listitem>
  *   </varlistentry>
  *   <varlistentry>
@@ -204,7 +204,7 @@ typedef gint (*WriteFunc) (GstAudioSink * sink, gpointer data, guint length);
 
 /* this internal thread does nothing else but write samples to the audio device.
  * It will write each segment in the ringbuffer and will update the play
- * pointer. 
+ * pointer.
  * The start/stop methods control the thread.
  */
 static void

gst-libs/gst/audio/gstaudiosrc.c

@@ -196,7 +196,7 @@ typedef guint (*ReadFunc)
 
 /* this internal thread does nothing else but read samples from the audio device.
  * It will read each segment in the ringbuffer and will update the play
- * pointer. 
+ * pointer.
  * The start/stop methods control the thread.
  */
 static void