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

15 files changed, 35 insertions, 35 deletions

diff --git a/gst/adder/gstadder.c b/gst/adder/gstadder.c
index 6afc92cbb..10fa22fd0 100644
--- a/gst/adder/gstadder.c
+++ b/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);
diff --git a/gst/audioconvert/audioconvert.c b/gst/audioconvert/audioconvert.c
index 8bea5efc6..1e42a2d56 100644
--- a/gst/audioconvert/audioconvert.c
+++ b/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;
diff --git a/gst/audiorate/gstaudiorate.c b/gst/audiorate/gstaudiorate.c
index 3e7d07686..8a3d6a53c 100644
--- a/gst/audiorate/gstaudiorate.c
+++ b/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,
diff --git a/gst/audioresample/gstaudioresample.c b/gst/audioresample/gstaudioresample.c
index 6def60157..67cc05d69 100644
--- a/gst/audioresample/gstaudioresample.c
+++ b/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);
 
diff --git a/gst/audioresample/resample.c b/gst/audioresample/resample.c
index 66c2c6dc5..fefa0c536 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 a7d337628..d37f15e21 100644
--- a/gst/encoding/gststreamsplitter.c
+++ b/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;
     }
diff --git a/gst/playback/gstdecodebin.c b/gst/playback/gstdecodebin.c
index 5104a4a0a..f4e5ba2d6 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 75de91959..137d27df0 100644
--- a/gst/playback/gstdecodebin2.c
+++ b/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 */
diff --git a/gst/playback/gstplaybin2.c b/gst/playback/gstplaybin2.c
index 269356474..4347e6a06 100644
--- a/gst/playback/gstplaybin2.c
+++ b/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,
diff --git a/gst/playback/gstplaysink.c b/gst/playback/gstplaysink.c
index 9f5335353..972c32383 100644
--- a/gst/playback/gstplaysink.c
+++ b/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");
diff --git a/gst/playback/gsturidecodebin.c b/gst/playback/gsturidecodebin.c
index 37cd834d3..5fc319e5f 100644
--- a/gst/playback/gsturidecodebin.c
+++ b/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.
diff --git a/gst/tcp/gstmultifdsink.c b/gst/tcp/gstmultifdsink.c
index e4e631021..5a5d29219 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
@@ -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;
diff --git a/gst/tcp/gsttcp.c b/gst/tcp/gsttcp.c
index d9f242c55..e5e7248b7 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 8c3f0e4e9..466950d03 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 f7cd51c97..c5db376a1 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.
  *