various: typo fixes

Fix typos in code and docs. Fixes. #658984

24 files changed, 63 insertions, 63 deletions

diff --git a/gst/adder/gstadder.c b/gst/adder/gstadder.c
index 3726f275a..9f6895ffc 100644
--- a/gst/adder/gstadder.c
+++ b/gst/adder/gstadder.c
@@ -1172,7 +1172,7 @@ 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_new_segment_full (FALSE, adder->segment_rate,
diff --git a/gst/audioconvert/audioconvert.c b/gst/audioconvert/audioconvert.c
index 524098c2a..d43432abd 100644
--- a/gst/audioconvert/audioconvert.c
+++ b/gst/audioconvert/audioconvert.c
@@ -280,7 +280,7 @@ 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. */
 
 /*** 
@@ -650,7 +650,7 @@ audio_convert_prepare_context (AudioConvertCtx * ctx, AudioConvertFmt * 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;
diff --git a/gst/audiorate/gstaudiorate.c b/gst/audiorate/gstaudiorate.c
index cf697c57d..4bf7d0af5 100644
--- a/gst/audiorate/gstaudiorate.c
+++ b/gst/audiorate/gstaudiorate.c
@@ -649,7 +649,7 @@ gst_audio_rate_chain (GstPad * pad, 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,
diff --git a/gst/audioresample/gstaudioresample.c b/gst/audioresample/gstaudioresample.c
index 80988cb7f..418a77c95 100644
--- a/gst/audioresample/gstaudioresample.c
+++ b/gst/audioresample/gstaudioresample.c
@@ -103,7 +103,7 @@ GST_STATIC_CAPS ( \
       "signed = (boolean) true" \
 )
 
-/* 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
@@ -187,7 +187,7 @@ gst_audio_resample_class_init (GstAudioResampleClass * klass)
    *
    * Length of the resample filter
    *
-   * Deprectated: Use #GstAudioResample:quality property instead
+   * Deprecated: Use #GstAudioResample:quality property instead
    */
   g_object_class_install_property (gobject_class, PROP_FILTER_LENGTH,
       g_param_spec_int ("filter-length", "Filter length",
@@ -1554,7 +1554,7 @@ _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);
 
diff --git a/gst/audioresample/resample.c b/gst/audioresample/resample.c
index 7cc04d66c..490eebcc4 100644
--- a/gst/audioresample/resample.c
+++ b/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) {
diff --git a/gst/encoding/gststreamsplitter.c b/gst/encoding/gststreamsplitter.c
index 53f755489..9221b353a 100644
--- a/gst/encoding/gststreamsplitter.c
+++ b/gst/encoding/gststreamsplitter.c
@@ -331,7 +331,7 @@ 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;
     }
diff --git a/gst/ffmpegcolorspace/avcodec.h b/gst/ffmpegcolorspace/avcodec.h
index 57f551ce5..6067aedb5 100644
--- a/gst/ffmpegcolorspace/avcodec.h
+++ b/gst/ffmpegcolorspace/avcodec.h
@@ -139,8 +139,8 @@ typedef struct AVCodecContext {
     /* video only */
     /**
      * frames per sec multiplied by frame_rate_base.
-     * for variable fps this is the precission, so if the timestamps 
-     * can be specified in msec precssion then this is 1000*frame_rate_base
+     * for variable fps this is the precision, so if the timestamps 
+     * can be specified in msec precision then this is 1000*frame_rate_base
      * - encoding: MUST be set by user
      * - decoding: set by lavc. 0 or the frame_rate if available
      */
diff --git a/gst/ffmpegcolorspace/gstffmpegcodecmap.c b/gst/ffmpegcolorspace/gstffmpegcodecmap.c
index 318a90ec0..97052cb45 100644
--- a/gst/ffmpegcolorspace/gstffmpegcodecmap.c
+++ b/gst/ffmpegcolorspace/gstffmpegcodecmap.c
@@ -151,7 +151,7 @@ gst_ff_aud_caps_new (AVCodecContext * context, const char *mimetype,
 }
 
 /* Convert a FFMPEG Pixel Format and optional AVCodecContext
- * to a GstCaps. If the context is ommitted, no fixed values
+ * to a GstCaps. If the context is omitted, no fixed values
  * for video/audio size will be included in the GstCaps
  *
  * See below for usefulness
@@ -453,7 +453,7 @@ gst_ffmpeg_pixfmt_to_caps (enum PixelFormat pix_fmt, AVCodecContext * context)
 }
 
 /* Convert a FFMPEG Sample Format and optional AVCodecContext
- * to a GstCaps. If the context is ommitted, no fixed values
+ * to a GstCaps. If the context is omitted, no fixed values
  * for video/audio size will be included in the GstCaps
  *
  * See below for usefulness
@@ -496,7 +496,7 @@ gst_ffmpeg_smpfmt_to_caps (enum SampleFormat sample_fmt,
 }
 
 /* Convert a FFMPEG codec Type and optional AVCodecContext
- * to a GstCaps. If the context is ommitted, no fixed values
+ * to a GstCaps. If the context is omitted, no fixed values
  * for video/audio size will be included in the GstCaps
  *
  * CodecType is primarily meant for uncompressed data GstCaps!
@@ -787,7 +787,7 @@ gst_ffmpeg_caps_to_pixfmt (const GstCaps * caps,
 }
 
 /* Convert a GstCaps and a FFMPEG codec Type to a
- * AVCodecContext. If the context is ommitted, no fixed values
+ * AVCodecContext. If the context is omitted, no fixed values
  * for video/audio size will be included in the context
  *
  * CodecType is primarily meant for uncompressed data GstCaps!
diff --git a/gst/ffmpegcolorspace/imgconvert.c b/gst/ffmpegcolorspace/imgconvert.c
index cb145bb5c..c670e25ff 100644
--- a/gst/ffmpegcolorspace/imgconvert.c
+++ b/gst/ffmpegcolorspace/imgconvert.c
@@ -1,5 +1,5 @@
 /*
- * Misc image convertion routines
+ * Misc image conversion routines
  * Copyright (c) 2001, 2002, 2003 Fabrice Bellard.
  *
  * This library is free software; you can redistribute it and/or
@@ -19,7 +19,7 @@
 
 /**
  * @file imgconvert.c
- * Misc image convertion routines.
+ * Misc image conversion routines.
  */
 
 /* TODO:
@@ -3079,7 +3079,7 @@ typedef struct ConvertEntry
       const AVPicture * src, int width, int height);
 } ConvertEntry;
 
-/* Add each new convertion function in this table. In order to be able
+/* Add each new conversion function in this table. In order to be able
    to convert from any format to any format, the following constraints
    must be satisfied:
 
diff --git a/gst/ffmpegcolorspace/imgconvert_template.h b/gst/ffmpegcolorspace/imgconvert_template.h
index 3b287e793..fbd5d4515 100644
--- a/gst/ffmpegcolorspace/imgconvert_template.h
+++ b/gst/ffmpegcolorspace/imgconvert_template.h
@@ -1,5 +1,5 @@
 /*
- * Templates for image convertion routines
+ * Templates for image conversion routines
  * Copyright (c) 2001, 2002, 2003 Fabrice Bellard.
  *
  * This library is free software; you can redistribute it and/or
diff --git a/gst/ffmpegcolorspace/mem.c b/gst/ffmpegcolorspace/mem.c
index 5c3a8a38e..fe1f0089d 100644
--- a/gst/ffmpegcolorspace/mem.c
+++ b/gst/ffmpegcolorspace/mem.c
@@ -111,7 +111,7 @@ av_realloc (void *ptr, unsigned int size)
 #endif
 }
 
-/* NOTE: ptr = NULL is explicetly allowed */
+/* NOTE: ptr = NULL is explictly allowed */
 void
 av_free (void *ptr)
 {
diff --git a/gst/playback/README b/gst/playback/README
index 286e49f81..8c5ef5006 100644
--- a/gst/playback/README
+++ b/gst/playback/README
@@ -48,7 +48,7 @@ playbasebin:
   is particulary important for chained oggs. Initially, a new group is created in 
   the 'building' state. All new streams will be added to the building group until
   no-more-pads is signaled or one of the preroll queues overflows. When this happens,
-  the group is commited to a list of groups ready for playback. PlaybaseBin will then
+  the group is committed to a list of groups ready for playback. PlaybaseBin will then
   attach a padprobe to each stream to figure out when it finished. It will remove
   the current group and install the next playable group, then.
 
@@ -73,7 +73,7 @@ playbin:
    stream detected. implements seeking and querying on the configured sinks.
 
    It also waits for new notifications from playbasebin about any new groups that are
-   becomming active. It then disconnects the sinks and reconnects them to the new
+   becoming active. It then disconnects the sinks and reconnects them to the new
    pads in the group.
 
    TODO
diff --git a/gst/playback/gstdecodebin.c b/gst/playback/gstdecodebin.c
index cab0542d9..7ce222fb4 100644
--- a/gst/playback/gstdecodebin.c
+++ b/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;
 
diff --git a/gst/playback/gstdecodebin2.c b/gst/playback/gstdecodebin2.c
index 8eff9de05..45ff2b14b 100644
--- a/gst/playback/gstdecodebin2.c
+++ b/gst/playback/gstdecodebin2.c
@@ -166,7 +166,7 @@ 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 */
@@ -688,7 +688,7 @@ gst_decode_bin_class_init (GstDecodeBinClass * 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. Decodebin2 will by default return an array with all
    * compatible factories, sorted by rank.
    *
@@ -722,7 +722,7 @@ gst_decode_bin_class_init (GstDecodeBinClass * klass)
    * @factories: A #GValueArray of possible #GstElementFactory to use.
    *
    * Once decodebin2 has found the possible #GstElementFactory objects to try
-   * for @caps on @pad, this signal is emited. The purpose of the signal is for
+   * for @caps on @pad, this signal is emitted. The purpose of the signal is for
    * the application to perform additional sorting or filtering on the element
    * factory array.
    *
@@ -755,7 +755,7 @@ gst_decode_bin_class_init (GstDecodeBinClass * klass)
    *
    * This signal is emitted once decodebin2 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 decodebin2 should do next.
@@ -856,7 +856,7 @@ gst_decode_bin_class_init (GstDecodeBinClass * klass)
   /**
    * GstDecodebin2: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
    */
@@ -868,7 +868,7 @@ gst_decode_bin_class_init (GstDecodeBinClass * klass)
   /**
    * GstDecodebin2: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
    */
@@ -880,7 +880,7 @@ gst_decode_bin_class_init (GstDecodeBinClass * klass)
   /**
    * GstDecodebin2: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
    */
@@ -3637,7 +3637,7 @@ 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 */
diff --git a/gst/playback/gstplaybasebin.c b/gst/playback/gstplaybasebin.c
index bee57f646..2d26aad88 100644
--- a/gst/playback/gstplaybasebin.c
+++ b/gst/playback/gstplaybasebin.c
@@ -705,7 +705,7 @@ queue_threshold_reached (GstElement * queue, GstPlayBaseBin * play_base_bin)
 /* this signal will be fired when one of the queues with raw
  * data is filled. This means that the group building stage is over
  * and playback of the new queued group should start. This is a rather unusual
- * situation because normally the group is commited when the "no_more_pads"
+ * situation because normally the group is committed when the "no_more_pads"
  * signal is fired.
  */
 static void
@@ -732,11 +732,11 @@ queue_out_of_data (GstElement * queue, GstPlayBaseBin * play_base_bin)
   GST_DEBUG_OBJECT (play_base_bin, "underrun signal received from queue %s",
       GST_ELEMENT_NAME (queue));
 
-  /* On underrun, we want to temoprarily pause playback, set a "min-size"
+  /* On underrun, we want to temporarily pause playback, set a "min-size"
    * threshold and wait for the running signal and then play again.
    *
    * This signal could never be called because the queue max-size limits are set
-   * too low. We take care of this possible deadlock in the the overrun signal
+   * too low. We take care of this possible deadlock in the overrun signal
    * handler. */
   g_signal_connect (G_OBJECT (queue), "pushing",
       G_CALLBACK (queue_threshold_reached), play_base_bin);
@@ -889,7 +889,7 @@ gen_preroll_element (GstPlayBaseBin * play_base_bin,
 
     gst_object_unref (sinkpad);
 
-    /* When we connect this queue, it will start running and immediatly
+    /* When we connect this queue, it will start running and immediately
      * fire an underrun. */
     g_signal_connect (G_OBJECT (preroll), "underrun",
         G_CALLBACK (queue_out_of_data), play_base_bin);
@@ -1894,7 +1894,7 @@ analyse_source (GstPlayBaseBin * play_base_bin, gboolean * is_raw,
         gst_iterator_resync (pads_iter);
         break;
       case GST_ITERATOR_OK:
-        /* we now officially have an ouput pad */
+        /* we now officially have an output pad */
         *have_out = TRUE;
 
         /* if FALSE, this pad has no caps and we continue with the next pad. */
diff --git a/gst/playback/gstplaybasebin.h b/gst/playback/gstplaybasebin.h
index c8c86499e..deceadf65 100644
--- a/gst/playback/gstplaybasebin.h
+++ b/gst/playback/gstplaybasebin.h
@@ -108,7 +108,7 @@ struct _GstPlayBaseBin {
 struct _GstPlayBaseBinClass {
   GstPipelineClass parent_class;
 
-  /* virtual fuctions */
+  /* virtual functions */
   gboolean (*setup_output_pads) (GstPlayBaseBin *play_base_bin,
                                  GstPlayBaseGroup *group);
 
diff --git a/gst/playback/gstplaybin.c b/gst/playback/gstplaybin.c
index 847246f0f..8597cff98 100644
--- a/gst/playback/gstplaybin.c
+++ b/gst/playback/gstplaybin.c
@@ -116,7 +116,7 @@
  * GNOME-based applications, for example, will usually want to create
  * gconfaudiosink and gconfvideosink elements and make playbin use those,
  * so that output happens to whatever the user has configured in the GNOME
- * Multimedia System Selector confinguration dialog.
+ * Multimedia System Selector configuration dialog.
  *
  * The sink elements do not necessarily need to be ready-made sinks. It is
  * possible to create container elements that look like a sink to playbin,
@@ -1207,7 +1207,7 @@ 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 output.  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
@@ -1519,7 +1519,7 @@ add_sink (GstPlayBin * play_bin, GstElement * sink, GstPad * srcpad,
     goto subtitle_failed;
 
 done:
-  /* we got the sink succesfully linked, now keep the sink
+  /* we got the sink successfully linked, now keep the sink
    * in our internal list */
   play_bin->sinks = g_list_prepend (play_bin->sinks, sink);
 
@@ -1791,7 +1791,7 @@ gst_play_bin_send_event_to_sink (GstPlayBin * play_bin, GstEvent * event)
     gst_event_ref (event);
     if ((res = gst_element_send_event (sink, event))) {
       GST_DEBUG_OBJECT (play_bin,
-          "Sent event succesfully to sink %" GST_PTR_FORMAT, sink);
+          "Sent event successfully to sink %" GST_PTR_FORMAT, sink);
       break;
     }
     GST_DEBUG_OBJECT (play_bin,
diff --git a/gst/playback/gstplaybin2.c b/gst/playback/gstplaybin2.c
index 74c5a1b20..9f86e3e3e 100644
--- a/gst/playback/gstplaybin2.c
+++ b/gst/playback/gstplaybin2.c
@@ -1128,7 +1128,7 @@ 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;
@@ -3114,7 +3114,7 @@ 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,
diff --git a/gst/playback/gstplaysink.c b/gst/playback/gstplaysink.c
index 8ab2eda0e..06d1081d0 100644
--- a/gst/playback/gstplaysink.c
+++ b/gst/playback/gstplaysink.c
@@ -3337,7 +3337,7 @@ 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");
     }
@@ -3346,7 +3346,7 @@ 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");
@@ -3354,7 +3354,7 @@ 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");
diff --git a/gst/playback/gsturidecodebin.c b/gst/playback/gsturidecodebin.c
index c672a3a52..e0660b5e2 100644
--- a/gst/playback/gsturidecodebin.c
+++ b/gst/playback/gsturidecodebin.c
@@ -105,7 +105,7 @@ 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 */
 
@@ -132,7 +132,7 @@ 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);
 };
 
@@ -513,7 +513,7 @@ 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.
    *
@@ -547,7 +547,7 @@ gst_uri_decode_bin_class_init (GstURIDecodeBinClass * klass)
    * @factories: A #GValueArray of possible #GstElementFactory to use.
    *
    * Once decodebin2 has found the possible #GstElementFactory objects to try
-   * for @caps on @pad, this signal is emited. The purpose of the signal is for
+   * for @caps on @pad, this signal is emitted. The purpose of the signal is for
    * the application to perform additional sorting or filtering on the element
    * factory array.
    *
@@ -582,7 +582,7 @@ 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 decodebin2 should do next.
@@ -1399,7 +1399,7 @@ analyse_source (GstURIDecodeBin * decoder, gboolean * is_raw,
         gst_iterator_resync (pads_iter);
         break;
       case GST_ITERATOR_OK:
-        /* we now officially have an ouput pad */
+        /* we now officially have an output pad */
         *have_out = TRUE;
 
         /* if FALSE, this pad has no caps and we continue with the next pad. */
diff --git a/gst/tcp/gstmultifdsink.c b/gst/tcp/gstmultifdsink.c
index b6c0f6d63..912c2738e 100644
--- a/gst/tcp/gstmultifdsink.c
+++ b/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
@@ -927,7 +927,7 @@ duplicate:
   }
 }
 
-/* "add" signal implemntation */
+/* "add" signal implementation */
 void
 gst_multi_fd_sink_add (GstMultiFdSink * sink, int fd)
 {
@@ -2248,7 +2248,7 @@ 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
@@ -2452,7 +2452,7 @@ 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;
diff --git a/gst/tcp/gsttcp.c b/gst/tcp/gsttcp.c
index 894fcb879..4479efb22 100644
--- a/gst/tcp/gsttcp.c
+++ b/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;
 }
 
diff --git a/gst/typefind/gsttypefindfunctions.c b/gst/typefind/gsttypefindfunctions.c
index 7784d2c87..d7541f47c 100644
--- a/gst/typefind/gsttypefindfunctions.c
+++ b/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;
diff --git a/gst/videotestsrc/gstvideotestsrc.c b/gst/videotestsrc/gstvideotestsrc.c
index 2dc01083b..2dbe17d25 100644
--- a/gst/videotestsrc/gstvideotestsrc.c
+++ b/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.
  *