diff --git a/gst-libs/gst/audio/audio.c b/gst-libs/gst/audio/audio.c index 9745abb4a9..de713d7b69 100644 --- a/gst-libs/gst/audio/audio.c +++ b/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. */ diff --git a/gst-libs/gst/audio/gstaudiobasesrc.c b/gst-libs/gst/audio/gstaudiobasesrc.c index f4890cfa52..0915348eb5 100644 --- a/gst-libs/gst/audio/gstaudiobasesrc.c +++ b/gst-libs/gst/audio/gstaudiobasesrc.c @@ -2,7 +2,7 @@ * Copyright (C) 1999,2000 Erik Walthinsen * 2005 Wim Taymans * - * 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. */ diff --git a/gst-libs/gst/audio/gstaudiocdsrc.c b/gst-libs/gst/audio/gstaudiocdsrc.c index 8296ab2c29..ebb6ecedcb 100644 --- a/gst-libs/gst/audio/gstaudiocdsrc.c +++ b/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 diff --git a/gst-libs/gst/audio/gstaudiodecoder.h b/gst-libs/gst/audio/gstaudiodecoder.h index 419f0251ac..8f97fe8beb 100644 --- a/gst-libs/gst/audio/gstaudiodecoder.h +++ b/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); diff --git a/gst-libs/gst/audio/gstaudioencoder.c b/gst-libs/gst/audio/gstaudioencoder.c index 52b25522ca..0b6f58757d 100644 --- a/gst-libs/gst/audio/gstaudioencoder.c +++ b/gst-libs/gst/audio/gstaudioencoder.c @@ -89,7 +89,7 @@ * * * 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 diff --git a/gst-libs/gst/audio/gstaudioringbuffer.c b/gst-libs/gst/audio/gstaudioringbuffer.c index da9ae4f183..ba56846b88 100644 --- a/gst-libs/gst/audio/gstaudioringbuffer.c +++ b/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. */ diff --git a/gst-libs/gst/audio/gstaudiosink.c b/gst-libs/gst/audio/gstaudiosink.c index 1c318e6902..66bb476082 100644 --- a/gst-libs/gst/audio/gstaudiosink.c +++ b/gst-libs/gst/audio/gstaudiosink.c @@ -47,7 +47,7 @@ * * * delay() - * Get the number of samples written but not yet played + * Get the number of samples written but not yet played * by the device. * * @@ -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 diff --git a/gst-libs/gst/audio/gstaudiosrc.c b/gst-libs/gst/audio/gstaudiosrc.c index fac7c9c020..71dc1e9fc1 100644 --- a/gst-libs/gst/audio/gstaudiosrc.c +++ b/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