Merge remote-tracking branch 'origin/master' into 0.11

Conflicts:
	gst-libs/gst/netbuffer/gstnetbuffer.c
	gst/ffmpegcolorspace/avcodec.h
	gst/ffmpegcolorspace/gstffmpegcodecmap.c
	gst/ffmpegcolorspace/imgconvert.c
	gst/ffmpegcolorspace/imgconvert_template.h
	gst/ffmpegcolorspace/mem.c
	gst/playback/README
	gst/playback/gstplaybasebin.c
	gst/playback/gstplaybasebin.h
	gst/playback/gstplaybin.c
	sys/v4l/v4lmjpegsrc_calls.c
	sys/v4l/videodev_mjpeg.h
	tests/check/elements/gnomevfssink.c

docs/design/design-audiosinks.txt

@@ -82,7 +82,7 @@ Design:
 
   Whenever new samples are to be put into the ringbuffer, the position of the
   read pointer is taken. The required write position is taken and the diff
-  is made between the required qnd actual position. If the defference is <0,
+  is made between the required and actual position. If the difference is <0,
   the sample is too late. If the difference is bigger than segtotal, the
   writing part has to wait for the play pointer to advance. 
 

docs/design/design-decodebin.txt

@@ -57,7 +57,7 @@ fine-tune the process.
 
     Get a list of elementfactories for @pad with @caps. This function is used to
     instruct decodebin2 of the elements it should try to autoplug. The default
-    behaviour when this function is not overridern is to get all elements that
+    behaviour when this function is not overriden is to get all elements that
     can handle @caps from the registry sorted by rank.
 
   - 'autoplug-select' :
@@ -142,7 +142,7 @@ Description:
 
   Multiple input-output data queue
   
-  The GstMultiQueue achieves the same functionnality as GstQueue, with a few
+  The GstMultiQueue achieves the same functionality as GstQueue, with a few
 differences:
 
   * Multiple streams handling.

docs/design/design-encoding.txt

@@ -16,13 +16,13 @@ A. Problems this proposal attempts to solve
 
 * Duplication of pipeline code for gstreamer-based applications
   wishing to encode and or mux streams, leading to subtle differences
-  and inconsistencies accross those applications.
+  and inconsistencies across those applications.
 
 * No unified system for describing encoding targets for applications
   in a user-friendly way.
 
 * No unified system for creating encoding targets for applications,
-  resulting in duplication of code accross all applications,
+  resulting in duplication of code across all applications,
   differences and inconsistencies that come with that duplication,
   and applications hardcoding element names and settings resulting in
   poor portability.

docs/design/design-orc-integration.txt

@@ -86,7 +86,7 @@ given an input format, channel position manipulation, dithering and
 quantizing configuration, and output format, a Orc code generator
 would create an OrcProgram, add the appropriate instructions to do
 each step based on the configuration, and then compile the program.
-Sucessfully compiling the program would return a function pointer
+Successfully compiling the program would return a function pointer
 that can be called to perform the operation.
 
 This sort of advanced usage requires structural changes to current

docs/design/draft-keyframe-force.txt

@@ -11,7 +11,7 @@ Consider the following use case:
   the existing file we are writing to and start writing to a new file.
 
   We want the new file to start with a keyframe so that one can start decoding
-  the file immediatly.
+  the file immediately.
 
 Components:
 

docs/design/draft-va.txt

@@ -7,7 +7,7 @@ Status:
   
 Purpose:
 
-  Provide an standarized generic way to introduce Video Acceleration APIs in
+  Provide an standardized generic way to introduce Video Acceleration APIs in
   already available elements instead of duplicating those into specialized ones.
   
   Provide a mechanism for a light GstBuffer subclassing in order to be able
@@ -26,7 +26,7 @@ Proposal:
 
    video/x-raw-va
    
-   Light subclassing embeding an structure in the data field of a standard
+   Light subclassing embedding an structure in the data field of a standard
    GstBuffer.
    
    struct {

ext/alsa/gstalsamixer.c

@@ -800,7 +800,7 @@ gst_alsa_mixer_set_record (GstAlsaMixer * mixer,
     snd_mixer_selem_set_capture_switch_all (alsa_track->element,
         record ? 1 : 0);
 
-    /* update all tracks in same exlusive cswitch group */
+    /* update all tracks in same exclusive cswitch group */
     if (alsa_track->alsa_flags & GST_ALSA_MIXER_TRACK_CSWITCH_EXCL) {
       GList *item;
 

ext/libvisual/visual.c

@@ -621,7 +621,7 @@ gst_visual_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
             GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
             GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
 
-        /* the max samples we must buffer buffer */
+        /* the max samples we must buffer */
         max_samples = MAX (VISUAL_SAMPLES, visual->spf);
         our_latency =
             gst_util_uint64_scale_int (max_samples, GST_SECOND,

ext/ogg/README

@@ -99,7 +99,7 @@ with great efficiency.
 1) the streaming mode.
 
 In this mode, the ogg demuxer receives buffers in the _chain() function which
-are then simply submited to the ogg sync layer. Pages are then processed when
+are then simply submitted to the ogg sync layer. Pages are then processed when
 the sync layer detects them, pads are created for new chains and packets are
 sent to the peer elements of the pads.
 

ext/ogg/gstoggdemux.c

@@ -549,7 +549,7 @@ gst_ogg_demux_chain_peer (GstOggPad * pad, ogg_packet * packet,
           pad->current_granule);
     } else if (ogg->segment.rate > 0.0 && pad->current_granule != -1) {
       pad->current_granule += duration;
-      GST_DEBUG_OBJECT (ogg, "interpollating granule %" G_GUINT64_FORMAT,
+      GST_DEBUG_OBJECT (ogg, "interpolating granule %" G_GUINT64_FORMAT,
           pad->current_granule);
     }
     if (ogg->segment.rate < 0.0 && packet->granulepos == -1) {

ext/theora/gsttheoradec.c

@@ -1509,7 +1509,7 @@ theora_dec_flush_decode (GstTheoraDec * dec)
   while (dec->queued) {
     GstBuffer *buf = GST_BUFFER_CAST (dec->queued->data);
 
-    /* iterate ouput queue an push downstream */
+    /* iterate output queue an push downstream */
     res = gst_pad_push (dec->srcpad, buf);
 
     dec->queued = g_list_delete_link (dec->queued, dec->queued);

ext/theora/gsttheoradec.h

@@ -76,7 +76,7 @@ struct _GstTheoraDec
   GstBufferPool *pool;
   gboolean use_cropping;
 
-  /* telemetry debuging options */
+  /* telemetry debugging options */
   gint telemetry_mv;
   gint telemetry_mbmode;
   gint telemetry_qi;

ext/theora/gsttheoraparse.c

@@ -328,7 +328,7 @@ theora_parse_set_streamheader (GstTheoraParse * parse)
   parse->shift = parse->info.keyframe_granule_shift;
 
   /* With libtheora-1.0beta1 the granulepos scheme was changed:
-   * where earlier the granulepos refered to the index/beginning
+   * where earlier the granulepos referred to the index/beginning
    * of a frame, it now refers to the end, which matches the use
    * in vorbis/speex. We check the bitstream version from the header so
    * we know which way to interpret the incoming granuepos

ext/vorbis/gstvorbisdec.c

@@ -534,7 +534,7 @@ vorbis_handle_data_packet (GstVorbisDec * vd, ogg_packet * packet,
   /* normal data packet */
   /* FIXME, we can skip decoding if the packet is outside of the
    * segment, this is however not very trivial as we need a previous
-   * packet to decode the current one so we must be carefull not to
+   * packet to decode the current one so we must be careful not to
    * throw away too much. For now we decode everything and clip right
    * before pushing data. */
 

gst-libs/gst/app/gstappsink.c

@@ -214,9 +214,9 @@ gst_app_sink_class_init (GstAppSinkClass * klass)
 
   /**
    * GstAppSink::eos:
-   * @appsink: the appsink element that emited the signal
+   * @appsink: the appsink element that emitted the signal
    *
-   * Signal that the end-of-stream has been reached. This signal is emited from
+   * Signal that the end-of-stream has been reached. This signal is emitted from
    * the steaming thread.
    */
   gst_app_sink_signals[SIGNAL_EOS] =
@@ -225,18 +225,18 @@ gst_app_sink_class_init (GstAppSinkClass * klass)
       NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
   /**
    * GstAppSink::new-preroll:
-   * @appsink: the appsink element that emited the signal
+   * @appsink: the appsink element that emitted the signal
    *
    * Signal that a new preroll sample is available.
    *
-   * This signal is emited from the steaming thread and only when the
+   * This signal is emitted from the steaming thread and only when the
    * "emit-signals" property is %TRUE.
    *
    * The new preroll sample can be retrieved with the "pull-preroll" action
    * signal or gst_app_sink_pull_preroll() either from this signal callback
    * or from any other thread.
    *
-   * Note that this signal is only emited when the "emit-signals" property is
+   * Note that this signal is only emitted when the "emit-signals" property is
    * set to %TRUE, which it is not by default for performance reasons.
    */
   gst_app_sink_signals[SIGNAL_NEW_PREROLL] =
@@ -249,14 +249,14 @@ gst_app_sink_class_init (GstAppSinkClass * klass)
    *
    * Signal that a new sample is available.
    *
-   * This signal is emited from the steaming thread and only when the
+   * This signal is emitted from the steaming thread and only when the
    * "emit-signals" property is %TRUE.
    *
    * The new sample can be retrieved with the "pull-sample" action
    * signal or gst_app_sink_pull_sample() either from this signal callback
    * or from any other thread.
    *
-   * Note that this signal is only emited when the "emit-signals" property is
+   * Note that this signal is only emitted when the "emit-signals" property is
    * set to %TRUE, which it is not by default for performance reasons.
    */
   gst_app_sink_signals[SIGNAL_NEW_SAMPLE] =
@@ -1211,7 +1211,7 @@ not_started:
  * This is an alternative to using the signals, it has lower overhead and is thus
  * less expensive, but also less flexible.
  *
- * If callbacks are installed, no signals will be emited for performance
+ * If callbacks are installed, no signals will be emitted for performance
  * reasons.
  *
  * Since: 0.10.23

gst-libs/gst/app/gstappsrc.c

@@ -37,7 +37,7 @@
  * byte buffers.
  *
  * The main way of handing data to the appsrc element is by calling the
- * gst_app_src_push_buffer() method or by emiting the push-buffer action signal.
+ * gst_app_src_push_buffer() method or by emitting the push-buffer action signal.
  * This will put the buffer onto a queue from which appsrc will read from in its
  * streaming thread. It is important to note that data transport will not happen
  * from the thread that performed the push-buffer call.
@@ -49,7 +49,7 @@
  * block the push-buffer method until free data becomes available again.
  *
  * When the internal queue is running out of data, the "need-data" signal is
- * emited, which signals the application that it should start pushing more data
+ * emitted, which signals the application that it should start pushing more data
  * into appsrc.
  *
  * In addition to the "need-data" and "enough-data" signals, appsrc can emit the
@@ -62,7 +62,7 @@
  * These signals allow the application to operate the appsrc in two different
  * ways:
  *
- * The push model, in which the application repeadedly calls the push-buffer method
+ * The push model, in which the application repeatedly calls the push-buffer method
  * with a new buffer. Optionally, the queue size in the appsrc can be controlled
  * with the enough-data and need-data signals by respectively stopping/starting
  * the push-buffer calls. This is a typical mode of operation for the
@@ -310,7 +310,7 @@ gst_app_src_class_init (GstAppSrcClass * klass)
   /**
    * GstAppSrc::block
    *
-   * When max-bytes are queued and after the enough-data signal has been emited,
+   * When max-bytes are queued and after the enough-data signal has been emitted,
    * block any further push-buffer calls until the amount of queued bytes drops
    * below the max-bytes limit.
    */
@@ -383,7 +383,7 @@ gst_app_src_class_init (GstAppSrcClass * klass)
 
   /**
    * GstAppSrc::need-data:
-   * @appsrc: the appsrc element that emited the signal
+   * @appsrc: the appsrc element that emitted the signal
    * @length: the amount of bytes needed.
    *
    * Signal that the source needs more data. In the callback or from another
@@ -402,11 +402,11 @@ gst_app_src_class_init (GstAppSrcClass * klass)
 
   /**
    * GstAppSrc::enough-data:
-   * @appsrc: the appsrc element that emited the signal
+   * @appsrc: the appsrc element that emitted the signal
    *
    * Signal that the source has enough data. It is recommended that the
    * application stops calling push-buffer until the need-data signal is
-   * emited again to avoid excessive buffer queueing.
+   * emitted again to avoid excessive buffer queueing.
    */
   gst_app_src_signals[SIGNAL_ENOUGH_DATA] =
       g_signal_new ("enough-data", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
@@ -415,7 +415,7 @@ gst_app_src_class_init (GstAppSrcClass * klass)
 
   /**
    * GstAppSrc::seek-data:
-   * @appsrc: the appsrc element that emited the signal
+   * @appsrc: the appsrc element that emitted the signal
    * @offset: the offset to seek to
    *
    * Seek to the given offset. The next push-buffer should produce buffers from
@@ -1008,7 +1008,7 @@ gst_app_src_create (GstBaseSrc * bsrc, guint64 offset, guint size,
        * random-access mode (where a buffer is normally pushed in the above
        * signal) we can still be empty because the pushed buffer got flushed or
        * when the application pushes the requested buffer later, we support both
-       * possiblities. */
+       * possibilities. */
       if (!g_queue_is_empty (priv->queue))
         continue;
 
@@ -1381,7 +1381,7 @@ gst_app_src_set_emit_signals (GstAppSrc * appsrc, gboolean emit)
  *
  * Check if appsrc will emit the "new-preroll" and "new-buffer" signals.
  *
- * Returns: %TRUE if @appsrc is emiting the "new-preroll" and "new-buffer"
+ * Returns: %TRUE if @appsrc is emitting the "new-preroll" and "new-buffer"
  * signals.
  *
  * Since: 0.10.23
@@ -1578,7 +1578,7 @@ flushing:
  * This is an alternative to using the signals, it has lower overhead and is thus
  * less expensive, but also less flexible.
  *
- * If callbacks are installed, no signals will be emited for performance
+ * If callbacks are installed, no signals will be emitted for performance
  * reasons.
  *
  * Since: 0.10.23

gst-libs/gst/app/gstappsrc.h

@@ -50,7 +50,7 @@ typedef struct _GstAppSrcPrivate GstAppSrcPrivate;
  *    and when it is set to -1, any number of bytes can be pushed into @appsrc.
  * @enough_data: Called when appsrc has enough data. It is recommended that the
  *    application stops calling push-buffer until the need_data callback is
- *    emited again to avoid excessive buffer queueing.
+ *    emitted again to avoid excessive buffer queueing.
  * @seek_data: Called when a seek should be performed to the offset.
  *    The next push-buffer should produce buffers from the new @offset.
  *    This callback is only called for seekable stream types.

gst-libs/gst/audio/audio.c

@@ -620,7 +620,7 @@ done:
  * @bpf: size of one audio frame in bytes. This is the size of one sample
  * * channels.
  *
- * Clip the the buffer to the given %GstSegment.
+ * Clip the buffer to the given %GstSegment.
  *
  * After calling this function the caller does not own a reference to
  * @buffer anymore.

gst-libs/gst/audio/gstaudiobasesink.c

@@ -2219,7 +2219,7 @@ gst_audio_base_sink_change_state (GstElement * element,
   /* ERRORS */
 open_failed:
   {
-    /* subclass must post a meaningfull error message */
+    /* subclass must post a meaningful error message */
     GST_DEBUG_OBJECT (sink, "open failed");
     return GST_STATE_CHANGE_FAILURE;
   }

gst-libs/gst/audio/gstaudiobasesrc.c

@@ -897,7 +897,7 @@ gst_audio_base_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
         running_time_sample =
             gst_util_uint64_scale_int (running_time, rate, GST_SECOND);
 
-        /* the segmentnr corrensponding to running_time, round down */
+        /* the segmentnr corresponding to running_time, round down */
         running_time_segment = running_time_sample / sps;
 
         /* the segment currently read from the ringbuffer */
@@ -923,7 +923,7 @@ gst_audio_base_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
          *
          * 1. We are more than the length of the ringbuffer behind.
          *    The length of the ringbuffer then gets to dictate
-         *    the threshold for what is concidered "too late"
+         *    the threshold for what is considered "too late"
          *
          * 2. If this is our first buffer.
          *    We know that we should catch up to running_time
@@ -1157,7 +1157,7 @@ gst_audio_base_src_change_state (GstElement * element,
   /* ERRORS */
 open_failed:
   {
-    /* subclass must post a meaningfull error message */
+    /* subclass must post a meaningful error message */
     GST_DEBUG_OBJECT (src, "open failed");
     return GST_STATE_CHANGE_FAILURE;
   }

gst-libs/gst/audio/gstaudioencoder.c

@@ -2007,7 +2007,7 @@ gst_audio_encoder_set_perfect_timestamp (GstAudioEncoder * enc,
  *
  * Queries encoder perfect timestamp behaviour.
  *
- * Returns: TRUE if pefect timestamp setting enabled.
+ * Returns: TRUE if perfect timestamp setting enabled.
  *
  * MT safe.
  *

gst-libs/gst/audio/gstaudioringbuffer.c

@@ -1501,7 +1501,7 @@ not_started:
  *
  * Commit @in_samples samples pointed to by @data to the ringbuffer @buf. 
  *
- * @in_samples and @out_samples define the rate conversion to perform on the the
+ * @in_samples and @out_samples define the rate conversion to perform on the
  * samples in @data. For negative rates, @out_samples must be negative and
  * @in_samples positive.
  *

gst-libs/gst/audio/multichannel.h

@@ -119,7 +119,7 @@ void    gst_audio_set_caps_channel_positions_list
                                          gint          num_positions);
 
 /* Custom fixate function. Elements that implement some sort of
- * channel conversion algorhithm should use this function for
+ * channel conversion algorithm should use this function for
  * fixating on GstAudioChannelPosition properties. It will take
  * care of equal channel positioning (left/right). Caller g_free()s
  * the return value. The input properties may be (and are supposed

gst-libs/gst/fft/gstfftf32.c

@@ -31,7 +31,7 @@
  *
  * #GstFFTF32 provides a FFT implementation and related functions for
  * 32 bit float samples. To use this call gst_fft_f32_new() for
- * allocating a #GstFFTF32 instance with the appropiate parameters and
+ * allocating a #GstFFTF32 instance with the appropriate parameters and
  * then call gst_fft_f32_fft() or gst_fft_f32_inverse_fft() to perform the
  * FFT or inverse FFT on a buffer of samples.
  *

gst-libs/gst/fft/gstfftf64.c

@@ -31,7 +31,7 @@
  *
  * #GstFFTF64 provides a FFT implementation and related functions for
  * 64 bit float samples. To use this call gst_fft_f64_new() for
- * allocating a #GstFFTF64 instance with the appropiate parameters and
+ * allocating a #GstFFTF64 instance with the appropriate parameters and
  * then call gst_fft_f64_fft() or gst_fft_f64_inverse_fft() to perform the
  * FFT or inverse FFT on a buffer of samples.
  *

gst-libs/gst/fft/gstffts16.c

@@ -31,7 +31,7 @@
  *
  * #GstFFTS16 provides a FFT implementation and related functions for
  * signed 16 bit integer samples. To use this call gst_fft_s16_new() for
- * allocating a #GstFFTS16 instance with the appropiate parameters and
+ * allocating a #GstFFTS16 instance with the appropriate parameters and
  * then call gst_fft_s16_fft() or gst_fft_s16_inverse_fft() to perform the
  * FFT or inverse FFT on a buffer of samples.
  *

gst-libs/gst/fft/gstffts32.c

@@ -31,7 +31,7 @@
  *
  * #GstFFTS32 provides a FFT implementation and related functions for
  * signed 32 bit integer samples. To use this call gst_fft_s32_new() for
- * allocating a #GstFFTS32 instance with the appropiate parameters and
+ * allocating a #GstFFTS32 instance with the appropriate parameters and
  * then call gst_fft_s32_fft() or gst_fft_s32_inverse_fft() to perform the
  * FFT or inverse FFT on a buffer of samples.
  *

gst-libs/gst/interfaces/navigation.c

@@ -53,7 +53,7 @@
  * mouse moving over a clickable region, or the set of available angles changing.
  * </para><para>
  * The GstNavigation message functions provide functions for creating and parsing
- * custom bus messages for signalling GstNavigation changes.
+ * custom bus messages for signaling GstNavigation changes.
  * </para>
  * </listitem>
  * </itemizedlist>

gst-libs/gst/pbutils/descriptions.c

@@ -155,7 +155,7 @@ static const FormatInfo formats[] = {
   {"video/sp5x", "Sunplus JPEG 5.x", 0},
   {"video/vivo", "Vivo", 0},
   {"video/x-3ivx", "3ivx", 0},
-  {"video/x-4xm", "4X Techologies Video", 0},
+  {"video/x-4xm", "4X Technologies Video", 0},
   {"video/x-apple-video", "Apple video", 0},
   {"video/x-aasc", "Autodesk Animator", 0},
   {"video/x-camtasia", "TechSmith Camtasia", 0},

gst-libs/gst/pbutils/encoding-profile.c

@@ -538,7 +538,7 @@ gst_encoding_video_profile_set_pass (GstEncodingVideoProfile * prof, guint pass)
  * @prof: a #GstEncodingVideoProfile
  * @variableframerate: a boolean
  *
- * If set to %TRUE, then the incoming streamm will be allowed to have non-constant
+ * If set to %TRUE, then the incoming stream will be allowed to have non-constant
  * framerate. If set to %FALSE (default value), then the incoming stream will
  * be normalized by dropping/duplicating frames in order to produce a
  * constance framerate.

gst-libs/gst/pbutils/encoding-target.h

@@ -36,7 +36,7 @@ G_BEGIN_DECLS
  * GST_ENCODING_CATEGORY_DEVICE:
  *
  * #GstEncodingTarget category for device-specific targets.
- * The name of the target will usually be the contructor and model of the device,
+ * The name of the target will usually be the constructor and model of the device,
  * and that target will contain #GstEncodingProfiles suitable for that device.
  */
 #define GST_ENCODING_CATEGORY_DEVICE            "device"

gst-libs/gst/pbutils/gstdiscoverer-types.c

@@ -1023,7 +1023,7 @@ DISCOVERER_INFO_ACCESSOR_CODE (duration, GstClockTime, GST_CLOCK_TIME_NONE);
  * gst_discoverer_info_get_seekable:
  * @info: a #GstDiscovererInfo
  *
- * Returns: the wheter the URI is seekable.
+ * Returns: the whether the URI is seekable.
  *
  * Since: 0.10.32
  */

gst-libs/gst/pbutils/gstdiscoverer.c

@@ -1469,7 +1469,7 @@ gst_discoverer_stop (GstDiscoverer * discoverer)
  * A copy of @uri will be made internally, so the caller can safely g_free()
  * afterwards.
  *
- * Returns: %TRUE if the @uri was succesfully appended to the list of pending
+ * Returns: %TRUE if the @uri was successfully appended to the list of pending
  * uris, else %FALSE
  *
  * Since: 0.10.31

gst-libs/gst/rtp/gstrtcpbuffer.c

@@ -634,7 +634,7 @@ gst_rtcp_packet_get_length (GstRTCPPacket * packet)
  * @ntptime: result NTP time
  * @rtptime: result RTP time
  * @packet_count: result packet count
- * @octet_count: result octect count
+ * @octet_count: result octet count
  *
  * Parse the SR sender info and store the values.
  */
@@ -675,7 +675,7 @@ gst_rtcp_packet_sr_get_sender_info (GstRTCPPacket * packet, guint32 * ssrc,
  * @ntptime: the NTP time
  * @rtptime: the RTP time
  * @packet_count: the packet count
- * @octet_count: the octect count
+ * @octet_count: the octet count
  *
  * Set the given values in the SR packet @packet.
  */
@@ -1158,7 +1158,7 @@ gst_rtcp_packet_sdes_next_entry (GstRTCPPacket * packet)
  *
  * When @type refers to a text item, @data will point to a UTF8 string. Note
  * that this UTF8 string is NOT null-terminated. Use
- * gst_rtcp_packet_sdes_copy_entry() to get a null-termined copy of the entry.
+ * gst_rtcp_packet_sdes_copy_entry() to get a null-terminated copy of the entry.
  *
  * Returns: %TRUE if there was valid data.
  */

gst-libs/gst/rtp/gstrtpbaseaudiopayload.c

@@ -848,7 +848,7 @@ gst_rtp_base_audio_payload_handle_buffer (GstRTPBasePayload *
         GstClockTime diff;
         guint64 bytes;
         /* we're only going to apply a positive gap, otherwise we let the marker
-         * bit do its thing. simply convert to bytes and add the the current
+         * bit do its thing. simply convert to bytes and add the current
          * offset */
         diff = timestamp - priv->last_timestamp;
         bytes = priv->time_to_bytes (payload, diff);

gst-libs/gst/rtp/gstrtpbuffer.c

@@ -323,7 +323,7 @@ validate_data (guint8 * data, guint len, guint8 * payload, guint payload_len)
     guint8 *extpos;
     guint16 extlen;
 
-    /* this points to the extenstion bits and header length */
+    /* this points to the extension bits and header length */
     extpos = &data[header_len];
 
     /* skip the header and check that we have enough space */

gst-libs/gst/rtsp/gstrtspconnection.c

@@ -1906,7 +1906,7 @@ build_next (GstRTSPBuilder * builder, GstRTSPMessage * message,
           goto done;
 
         /* we have the complete body now, store in the message adjusting the
-         * length to include the traling '\0' */
+         * length to include the trailing '\0' */
         gst_rtsp_message_take_body (message,
             (guint8 *) builder->body_data, builder->body_len + 1);
         builder->body_data = NULL;

gst-libs/gst/rtsp/gstrtsprange.c

@@ -263,7 +263,7 @@ gst_rtsp_range_to_string (const GstRTSPTimeRange * range)
  * gst_rtsp_range_free:
  * @range: a #GstRTSPTimeRange
  *
- * Free the memory alocated by @range.
+ * Free the memory allocated by @range.
  */
 void
 gst_rtsp_range_free (GstRTSPTimeRange * range)

gst-libs/gst/tag/gstexiftag.c

@@ -1549,7 +1549,7 @@ write_exif_ifd (const GstTagList * taglist, gboolean byte_order,
   else
     gst_byte_writer_put_uint16_be (&writer.tagwriter, writer.tags_total);
 
-  GST_DEBUG ("Number of tags rewriten to %d", writer.tags_total);
+  GST_DEBUG ("Number of tags rewritten to %d", writer.tags_total);
 
   /* now that we know the tag headers size, we can add the offsets */
   gst_exif_tag_rewrite_offsets (&writer.tagwriter, writer.byte_order,
@@ -2057,7 +2057,7 @@ deserialize_geo_coordinate (GstExifReader * exif_reader,
   }
 
   if (exiftag->exif_tag != next_tagdata.tag) {
-    GST_WARNING ("This is not a geo cordinate tag");
+    GST_WARNING ("This is not a geo coordinate tag");
     return ret;
   }
 

gst-libs/gst/tag/gstvorbistag.c

@@ -647,7 +647,7 @@ gst_tag_to_metadata_block_picture (const gchar * tag,
  * Creates a new tag list that contains the information parsed out of a
  * vorbiscomment packet.
  *
- * Returns: A #GList of newly-allowcated key=value strings. Free with
+ * Returns: A #GList of newly-allocated key=value strings. Free with
  *          g_list_foreach (list, (GFunc) g_free, NULL) plus g_list_free (list)
  */
 GList *

gst-libs/gst/tag/gstxmptag.c

@@ -1411,7 +1411,7 @@ gst_tag_list_from_xmp_buffer (GstBuffer * buffer)
                 }
               } else {
                 XmpTag *xmp_tag = NULL;
-                /* FIXME: eventualy rewrite ns
+                /* FIXME: eventually rewrite ns
                  * find ':'
                  * check if ns before ':' is in ns_map and ns_map[i].gstreamer_ns!=NULL
                  * do 2 stage filter in tag_matches
@@ -1467,7 +1467,7 @@ gst_tag_list_from_xmp_buffer (GstBuffer * buffer)
              <dc:type><rdf:Bag><rdf:li>Image</rdf:li></rdf:Bag></dc:type>
              <dc:creator><rdf:Seq><rdf:li/></rdf:Seq></dc:creator>
            */
-          /* FIXME: eventualy rewrite ns */
+          /* FIXME: eventually rewrite ns */
 
           /* skip rdf tags for now */
           if (strncmp (part, "rdf:", 4)) {
@@ -1852,7 +1852,7 @@ gst_tag_list_to_xmp_buffer_full (const GstTagList * list, gboolean read_only,
   g_string_append (data, "</x:xmpmeta>\n");
 
   if (!read_only) {
-    /* the xmp spec recommand to add 2-4KB padding for in-place editable xmp */
+    /* the xmp spec recommends to add 2-4KB padding for in-place editable xmp */
     guint i;
 
     for (i = 0; i < 32; i++) {

gst-libs/gst/tag/id3v2.3.0.txt

@@ -183,7 +183,7 @@ bits are ignored, so a 257 bytes long tag is represented as $00 00 02 01.
 The ID3v2 tag size is the size of the complete tag after unsychronisation,
 including padding, excluding the header but not excluding the extended header
 (total tag size - 10). Only 28 bits (representing up to 256MB) are used in the
-size description to avoid the introducuction of 'false syncsignals'.
+size description to avoid the introduction of 'false syncsignals'.
 An ID3v2 tag can be detected with the following pattern:
 
   $49 44 33 yy yy xx zz zz zz zz
@@ -1006,7 +1006,7 @@ Where time stamp format is:
   $01 Absolute time, 32 bit sized, using MPEG frames as unit
   $02 Absolute time, 32 bit sized, using milliseconds as unit
 
-Abolute time means that every stamp contains the time from the beginning of the
+Absolute time means that every stamp contains the time from the beginning of the
 file.
 Followed by a list of key events in the following format:
 
@@ -1111,7 +1111,7 @@ Where time stamp format is:
   $01 Absolute time, 32 bit sized, using MPEG frames as unit
   $02 Absolute time, 32 bit sized, using milliseconds as unit
 
-Abolute time means that every stamp contains the time from the beginning of the
+Absolute time means that every stamp contains the time from the beginning of the
 file.
 
 4.9. Unsychronised lyrics/text transcription
@@ -1167,7 +1167,7 @@ Time stamp format is:
   $01 Absolute time, 32 bit sized, using MPEG frames as unit
   $02 Absolute time, 32 bit sized, using milliseconds as unit
 
-Abolute time means that every stamp contains the time from the beginning of the
+Absolute time means that every stamp contains the time from the beginning of the
 file.
 The text that follows the frame header differs from that of the unsynchronised
 lyrics/text transcription in one major way. Each syllable (or whatever size of
@@ -1463,7 +1463,7 @@ frame in each tag.
 4.20. Audio encryption
 
 This frame indicates if the actual audio stream is encrypted, and by whom.
-Since standardisation of such encrypion scheme is beyond this document, all
+Since standardisation of such encryption scheme is beyond this document, all
 "AENC" frames begin with a terminated string with a URL containing an email
 address, or a link to a location where an email address can be found, that
 belongs to the organisation responsible for this specific encrypted audio file.

gst-libs/gst/tag/id3v2.4.0-frames.txt

@@ -255,7 +255,7 @@ Abstract
    one text information frame of its kind in an tag. All text
    information frames supports multiple strings, stored as a null
    separated list, where null is reperesented by the termination code
-   for the charater encoding. All text frame identifiers begin with "T".
+   for the character encoding. All text frame identifiers begin with "T".
    Only text frame identifiers begin with "T", with the exception of the
    "TXXX" frame. All the text information frames have the following
    format:

gst-libs/gst/tag/id3v2.4.0-structure.txt

@@ -411,7 +411,7 @@ Abstract
    byte indicates that extra information is added to the header. These
    fields of extra information is ordered as the flags that indicates
    them. The flags field is defined as follows (l and o left out because
-   ther resemblence to one and zero):
+   their resemblence to one and zero):
 
      %0abc0000 %0h00kmnp
 

gst-libs/gst/video/videooverlay.c

@@ -423,7 +423,7 @@ gst_video_overlay_expose (GstVideoOverlay * overlay)
  * @handle_events: a #gboolean indicating if events should be handled or not.
  *
  * Tell an overlay that it should handle events from the window system. These
- * events are forwared upstream as navigation events. In some window system,
+ * events are forwarded upstream as navigation events. In some window system,
  * events are not propagated in the window hierarchy if a client is listening
  * for them. This method allows you to disable events handling completely
  * from the XOverlay.

gst/adder/gstadder.c

@@ -1193,7 +1193,7 @@ gst_adder_collected (GstCollectPads * pads, gpointer user_data)
      * - currently we just set rate as received from last seek-event
      *
      * When seeking we set the start and stop positions as given in the seek
-     * event. We also adjust offset & timestamp acordingly.
+     * event. We also adjust offset & timestamp accordingly.
      * This basically ignores all newsegments sent by upstream.
      */
     event = gst_event_new_segment (&adder->segment);

gst/audioconvert/audioconvert.c

@@ -279,7 +279,7 @@ MAKE_UNPACK_FUNC_ORC_IF (s32_le_float, 4, 0, READ32_FROM_LE);
 MAKE_UNPACK_FUNC_ORC_IF (u32_be_float, 4, SIGNED, READ32_FROM_BE);
 MAKE_UNPACK_FUNC_ORC_IF (s32_be_float, 4, 0, READ32_FROM_BE);
 
-/* One of the double_hq_* functions generated above is ineffecient, but it's
+/* One of the double_hq_* functions generated above is inefficient, but it's
  * never used anyway.  The same is true for one of the s32_* functions. */
 
 /*** 
@@ -640,7 +640,7 @@ audio_convert_prepare_context (AudioConvertCtx * ctx, GstAudioInfo * in,
   ctx->pack = pack_funcs[idx_out];
 
   /* if both formats are float/double or we use noise shaping use double as
-   * intermediate format and and switch mixing */
+   * intermediate format and switch mixing */
   if (!DOUBLE_INTERMEDIATE_FORMAT (ctx)) {
     GST_INFO ("use int mixing");
     ctx->channel_mix = (AudioConvertMix) gst_channel_mix_mix_int;

gst/audiorate/gstaudiorate.c

@@ -544,7 +544,7 @@ gst_audio_rate_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
       GST_BUFFER_OFFSET_END (fill) = audiorate->next_offset;
 
       /* Use next timestamp, then calculate following timestamp based on 
-       * offset to get duration. Neccesary complexity to get 'perfect' 
+       * offset to get duration. Necessary complexity to get 'perfect' 
        * streams */
       GST_BUFFER_TIMESTAMP (fill) = audiorate->next_ts;
       audiorate->next_ts = gst_util_uint64_scale_int (audiorate->next_offset,

gst/audioresample/gstaudioresample.c

@@ -73,7 +73,7 @@ enum
   GST_AUDIO_CAPS_MAKE ("{ F32BE, F64BE, S32BE, S24BE, S16BE, S8 }")
 #endif
 
-/* If TRUE integer arithmetic resampling is faster and will be used if appropiate */
+/* If TRUE integer arithmetic resampling is faster and will be used if appropriate */
 #if defined AUDIORESAMPLE_FORMAT_INT
 static gboolean gst_audio_resample_use_int = TRUE;
 #elif defined AUDIORESAMPLE_FORMAT_FLOAT
@@ -1395,7 +1395,7 @@ _benchmark_integer_resampling (void)
   resample_int_resampler_destroy (stb);
 
   if (av > bv)
-    GST_INFO ("Using integer resampler if appropiate: %lf < %lf", bv, av);
+    GST_INFO ("Using integer resampler if appropriate: %lf < %lf", bv, av);
   else
     GST_INFO ("Using float resampler for everything: %lf <= %lf", av, bv);
 

gst/audioresample/resample.c

@@ -461,7 +461,7 @@ resampler_basic_direct_single (SpeexResamplerState * st,
       sum += MULT16_16 (sinc[j], iptr[j]);
 
 /*    This code is slower on most DSPs which have only 2 accumulators.
-      Plus this this forces truncation to 32 bits and you lose the HW guard bits.
+      Plus this forces truncation to 32 bits and you lose the HW guard bits.
       I think we can trust the compiler and let it vectorize and/or unroll itself.
       spx_word32_t accum[4] = {0,0,0,0};
       for(j=0;j<N;j+=4) {

gst/encoding/gststreamsplitter.c

@@ -367,7 +367,7 @@ resync:
 
     if (res) {
       /* FIXME : we need to switch properly */
-      GST_DEBUG_OBJECT (srcpad, "Setting caps on this pad was succesfull");
+      GST_DEBUG_OBJECT (srcpad, "Setting caps on this pad was successful");
       stream_splitter->current = srcpad;
       goto beach;
     }

gst/playback/gstdecodebin.c

@@ -1441,7 +1441,7 @@ 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.
-   * Note: be very carefull with thread safety here as this underrun
+   * Note: be very careful 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
    * list) and the overrun signals that are signalled from the
@@ -1773,7 +1773,7 @@ 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 careful not to signal the
    * no_more_pads signal after connecting the first pad. */
   more = g_list_length (to_connect) > 1;
 

gst/playback/gstdecodebin2.c

@@ -165,7 +165,7 @@ struct _GstDecodeBin
   gboolean have_type;           /* if we received the have_type signal */
   guint have_type_id;           /* signal id for have-type from typefind */
 
-  gboolean async_pending;       /* async-start has been emited */
+  gboolean async_pending;       /* async-start has been emitted */
 
   GMutex *dyn_lock;             /* lock protecting pad blocking */
   gboolean shutdown;            /* if we are shutting down */
@@ -716,7 +716,7 @@ gst_decode_bin_class_init (GstDecodeBinClass * klass)
    *
    * This signal is emitted once decodebin has found all the possible
    * #GstElementFactory that can be used to handle the given @caps. For each of
-   * those factories, this signal is emited.
+   * those factories, this signal is emitted.
    *
    * The signal handler should return a #GST_TYPE_AUTOPLUG_SELECT_RESULT enum
    * value indicating what decodebin should do next.
@@ -817,7 +817,7 @@ gst_decode_bin_class_init (GstDecodeBinClass * klass)
   /**
    * GstDecodeBin:max-size-bytes
    *
-   * Max amount amount of bytes in the queue (0=automatic).
+   * Max amount of bytes in the queue (0=automatic).
    *
    * Since: 0.10.26
    */
@@ -829,7 +829,7 @@ gst_decode_bin_class_init (GstDecodeBinClass * klass)
   /**
    * GstDecodeBin:max-size-buffers
    *
-   * Max amount amount of buffers in the queue (0=automatic).
+   * Max amount of buffers in the queue (0=automatic).
    *
    * Since: 0.10.26
    */
@@ -841,7 +841,7 @@ gst_decode_bin_class_init (GstDecodeBinClass * klass)
   /**
    * GstDecodeBin:max-size-time
    *
-   * Max amount amount of time in the queue (in ns, 0=automatic).
+   * Max amount of time in the queue (in ns, 0=automatic).
    *
    * Since: 0.10.26
    */
@@ -3632,7 +3632,7 @@ gst_decode_bin_expose (GstDecodeBin * dbin)
 
   /* 4. Signal no-more-pads. This allows the application to hook stuff to the
    * exposed pads */
-  GST_LOG_OBJECT (dbin, "signalling no-more-pads");
+  GST_LOG_OBJECT (dbin, "signaling no-more-pads");
   gst_element_no_more_pads (GST_ELEMENT (dbin));
 
   /* 5. Send a custom element message with the stream topology */

gst/playback/gstplaybin2.c

@@ -1129,7 +1129,7 @@ init_group (GstPlayBin * playbin, GstSourceGroup * group)
    * matches the media. */
   group->playbin = playbin;
   /* If you add any items to these lists, check that media_list[] is defined
-   * above to be large enough to hold MAX(items)+1, so as to accomodate a
+   * above to be large enough to hold MAX(items)+1, so as to accommodate a
    * NULL terminator (set when the memory is zeroed on allocation) */
   group->selector[PLAYBIN_STREAM_AUDIO].media_list[0] = "audio/";
   group->selector[PLAYBIN_STREAM_AUDIO].type = GST_PLAY_SINK_TYPE_AUDIO;
@@ -3124,7 +3124,7 @@ autoplug_factories_cb (GstElement * decodebin, GstPad * pad,
  * supported subtitles directly */
 
 /* FIXME 0.11: Remove the checks for ANY caps, a sink should specify
- * explicitely the caps it supports and if it claims to support ANY
+ * explicitly the caps it supports and if it claims to support ANY
  * caps it really should support everything */
 static gboolean
 autoplug_continue_cb (GstElement * element, GstPad * pad, GstCaps * caps,

gst/playback/gstplaysink.c

@@ -3367,7 +3367,7 @@ gst_play_sink_send_event_to_sink (GstPlaySink * playsink, GstEvent * event)
   if (playsink->textchain && playsink->textchain->sink) {
     gst_event_ref (event);
     if ((res = gst_element_send_event (playsink->textchain->chain.bin, event))) {
-      GST_DEBUG_OBJECT (playsink, "Sent event succesfully to text sink");
+      GST_DEBUG_OBJECT (playsink, "Sent event successfully to text sink");
     } else {
       GST_DEBUG_OBJECT (playsink, "Event failed when sent to text sink");
     }
@@ -3376,7 +3376,7 @@ gst_play_sink_send_event_to_sink (GstPlaySink * playsink, GstEvent * event)
   if (playsink->videochain) {
     gst_event_ref (event);
     if ((res = gst_element_send_event (playsink->videochain->chain.bin, event))) {
-      GST_DEBUG_OBJECT (playsink, "Sent event succesfully to video sink");
+      GST_DEBUG_OBJECT (playsink, "Sent event successfully to video sink");
       goto done;
     }
     GST_DEBUG_OBJECT (playsink, "Event failed when sent to video sink");
@@ -3384,7 +3384,7 @@ gst_play_sink_send_event_to_sink (GstPlaySink * playsink, GstEvent * event)
   if (playsink->audiochain) {
     gst_event_ref (event);
     if ((res = gst_element_send_event (playsink->audiochain->chain.bin, event))) {
-      GST_DEBUG_OBJECT (playsink, "Sent event succesfully to audio sink");
+      GST_DEBUG_OBJECT (playsink, "Sent event successfully to audio sink");
       goto done;
     }
     GST_DEBUG_OBJECT (playsink, "Event failed when sent to audio sink");

gst/playback/gsturidecodebin.c

@@ -106,7 +106,7 @@ struct _GstURIDecodeBin
   guint src_nmp_sig_id;         /* no-more-pads signal id */
   gint pending;
 
-  gboolean async_pending;       /* async-start has been emited */
+  gboolean async_pending;       /* async-start has been emitted */
 
   gboolean expose_allstreams;   /* Whether to expose unknow type streams or not */
 
@@ -133,7 +133,7 @@ struct _GstURIDecodeBinClass
     GstAutoplugSelectResult (*autoplug_select) (GstElement * element,
       GstPad * pad, GstCaps * caps, GstElementFactory * factory);
 
-  /* emited when all data is decoded */
+  /* emitted when all data is decoded */
   void (*drained) (GstElement * element);
 };
 
@@ -502,7 +502,7 @@ gst_uri_decode_bin_class_init (GstURIDecodeBinClass * klass)
    * @pad: The #GstPad.
    * @caps: The #GstCaps found.
    *
-   * This function is emited when an array of possible factories for @caps on
+   * This function is emitted when an array of possible factories for @caps on
    * @pad is needed. Uridecodebin will by default return an array with all
    * compatible factories, sorted by rank.
    *
@@ -571,7 +571,7 @@ gst_uri_decode_bin_class_init (GstURIDecodeBinClass * klass)
    *
    * This signal is emitted once uridecodebin has found all the possible
    * #GstElementFactory that can be used to handle the given @caps. For each of
-   * those factories, this signal is emited.
+   * those factories, this signal is emitted.
    *
    * The signal handler should return a #GST_TYPE_AUTOPLUG_SELECT_RESULT enum
    * value indicating what decodebin should do next.

gst/tcp/gstmultifdsink.c

@@ -67,7 +67,7 @@
  * prefer a minimum burst size even if it requires not starting with a keyframe.
  *
  * Multifdsink can be instructed to keep at least a minimum amount of data
- * expressed in time or byte units in its internal queues with the the 
+ * expressed in time or byte units in its internal queues with the 
  * #GstMultiFdSink:time-min and #GstMultiFdSink:bytes-min properties respectively.
  * These properties are useful if the application adds clients with the 
  * #GstMultiFdSink::add-full signal to make sure that a burst connect can
@@ -913,7 +913,7 @@ duplicate:
   }
 }
 
-/* "add" signal implemntation */
+/* "add" signal implementation */
 void
 gst_multi_fd_sink_add (GstMultiFdSink * sink, int fd)
 {
@@ -2126,7 +2126,7 @@ gst_multi_fd_sink_recover_client (GstMultiFdSink * sink, GstTCPClient * client)
  *
  * Special care is taken of clients that were waiting for a new buffer (they
  * had a position of -1) because they can proceed after adding this new buffer.
- * This is done by adding the client back into the write fd_set and signalling
+ * This is done by adding the client back into the write fd_set and signaling
  * the select thread that the fd_set changed.
  */
 static void
@@ -2330,7 +2330,7 @@ gst_multi_fd_sink_handle_clients (GstMultiFdSink * sink)
         GST_CLOCK_TIME_NONE);
 
     /* Handle the special case in which the sink is not receiving more buffers
-     * and will not disconnect innactive client in the streaming thread. */
+     * and will not disconnect inactive client in the streaming thread. */
     if (G_UNLIKELY (result == 0)) {
       GstClockTime now;
       GTimeVal nowtv;

gst/tcp/gsttcp.c

@@ -116,7 +116,7 @@ gst_tcp_socket_write (int socket, const void *buf, size_t count)
     bytes_written += wrote;
   }
 
-  GST_LOG ("wrote %" G_GSIZE_FORMAT " bytes succesfully", bytes_written);
+  GST_LOG ("wrote %" G_GSIZE_FORMAT " bytes successfully", bytes_written);
   return bytes_written;
 }
 

gst/typefind/gsttypefindfunctions.c

@@ -1061,7 +1061,7 @@ mp3_type_frame_length_from_header (guint32 header, guint * put_layer,
   /* bitrate index */
   bitrate = header & 0xF;
   if (bitrate == 0 && possible_free_framelen == -1) {
-    GST_LOG ("Possibly a free format mp3 - signalling");
+    GST_LOG ("Possibly a free format mp3 - signaling");
     *may_be_free_format = TRUE;
   }
   if (bitrate == 15 || (bitrate == 0 && possible_free_framelen == -1))
@@ -1440,7 +1440,7 @@ ac3_type_find (GstTypeFind * tf, gpointer unused)
 {
   DataScanCtx c = { 0, NULL, 0 };
 
-  /* Search for an ac3 frame; not neccesarily right at the start, but give it
+  /* Search for an ac3 frame; not necessarily right at the start, but give it
    * a lower probability if not found right at the start. Check that the
    * frame is followed by a second frame at the expected offset.
    * We could also check the two ac3 CRCs, but we don't do that right now */
@@ -1607,7 +1607,7 @@ dts_type_find (GstTypeFind * tf, gpointer unused)
 {
   DataScanCtx c = { 0, NULL, 0 };
 
-  /* Search for an dts frame; not neccesarily right at the start, but give it
+  /* Search for an dts frame; not necessarily right at the start, but give it
    * a lower probability if not found right at the start. Check that the
    * frame is followed by a second frame at the expected offset. */
   while (c.offset <= DTS_MAX_FRAMESIZE) {
@@ -2412,7 +2412,7 @@ h264_video_type_find (GstTypeFind * tf, gpointer unused)
       nut = c.data[3] & 0x9f;   /* forbiden_zero_bit | nal_unit_type */
       ref = c.data[3] & 0x60;   /* nal_ref_idc */
 
-      /* if forbiden bit is different to 0 won't be h264 */
+      /* if forbidden bit is different to 0 won't be h264 */
       if (nut > 0x1f) {
         bad++;
         break;

gst/videotestsrc/gstvideotestsrc.c

@@ -21,7 +21,7 @@
 /**
  * SECTION:element-videotestsrc
  *
- * The videotestsrc element is used to produce test video data in a wide variaty
+ * The videotestsrc element is used to produce test video data in a wide variety
  * of formats. The video test data produced can be controlled with the "pattern"
  * property.
  *

m4/freetype2.m4

@@ -131,7 +131,7 @@ else
        echo "*** The FreeType test program failed to run.  If your system uses"
        echo "*** shared libraries and they are installed outside the normal"
        echo "*** system library path, make sure the variable LD_LIBRARY_PATH"
-       echo "*** (or whatever is appropiate for your system) is correctly set."
+       echo "*** (or whatever is appropriate for your system) is correctly set."
      fi
    fi
    FT2_CFLAGS=""

sys/xvimage/xvimagesink.c

@@ -22,7 +22,7 @@
  * SECTION:element-xvimagesink
  *
  * XvImageSink renders video frames to a drawable (XWindow) on a local display
- * using the XVideo extension. Rendering to a remote display is theorically
+ * using the XVideo extension. Rendering to a remote display is theoretically
  * possible but i doubt that the XVideo extension is actually available when
  * connecting to a remote display. This element can receive a Window ID from the
  * application through the XOverlay interface and will then render video frames

sys/xvimage/xvimagesink.h

@@ -262,7 +262,7 @@ struct _GstXvImageSink
   /* stream metadata */
   gchar *media_title;
 
-  /* target video rectagle */
+  /* target video rectangle */
   GstVideoRectangle render_rect;
   gboolean have_render_rect;
 };

tests/check/elements/adder.c

@@ -534,7 +534,7 @@ GST_START_TEST (test_live_seeking)
 #if 1
     fail_unless (res == TRUE, NULL);
 #else
-    /* adder is picky, if a single seek fails it totaly fails */
+    /* adder is picky, if a single seek fails it totally fails */
     fail_unless (res == FALSE, NULL);
 #endif
 

tests/check/elements/audioresample.c

@@ -793,7 +793,7 @@ fakesink_handoff_cb (GstElement * object, GstBuffer * buffer, GstPad * pad,
     ctx->latency = 1000 - gst_buffer_get_size (buffer) / 8;
   }
 
-  /* Check if we have a perfectly timestampped stream */
+  /* Check if we have a perfectly timestamped stream */
   if (ctx->next_out_ts != GST_CLOCK_TIME_NONE)
     fail_unless (ctx->next_out_ts == GST_BUFFER_TIMESTAMP (buffer),
         "expected timestamp %" GST_TIME_FORMAT " got timestamp %"

tests/check/elements/textoverlay.c

@@ -362,7 +362,7 @@ GST_START_TEST (test_video_passthrough)
   /* pushing gives away one of the two references we have ... */
   fail_unless (gst_pad_push (myvideosrcpad, inbuffer) == GST_FLOW_OK);
 
-  /* should have been discareded as out-of-segment since it has no timestamp */
+  /* should have been discarded as out-of-segment since it has no timestamp */
   ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
   fail_unless_equals_int (g_list_length (buffers), 0);
 

tests/examples/encoding/encoding.c

@@ -501,7 +501,7 @@ main (int argc, char **argv)
       return 1;
     }
 
-    /* Trancode file */
+    /* Transcode file */
     transcode_file (inputuri, outputuri, prof);
 
     /* cleanup */