documentation: fix some typos

NEWS

@@ -353,7 +353,7 @@ New element features and additions
 
 -   rtpjitterbuffer has improved end-of-stream handling
 
--   rtpmp4vpay will be prefered over rtpmp4gpay for MPEG-4 video in
+-   rtpmp4vpay will be preferred over rtpmp4gpay for MPEG-4 video in
     autoplugging scenarios now
 
 -   rtspsrc now allows applications to send RTSP SET_PARAMETER and
@@ -1208,7 +1208,7 @@ Cerbero has seen a number of improvements:
     used in order to re-produce a specific build. To set a manifest, you
     can set manifest = 'my_manifest.xml' in your configuration file, or
     use the --manifest command line option. The command line option will
-    take precendence over anything specific in the configuration file.
+    take precedence over anything specific in the configuration file.
 
 -   The new build-deps command can be used to build only the
     dependencies of a recipe, without the recipe itself.

ext/mpeg2dec/gstmpeg2dec.c

@@ -729,7 +729,7 @@ handle_sequence (GstMpeg2dec * mpeg2dec, const mpeg2_info_t * info)
         /* 0 forbidden */
         /* 2 unspecified */
         /* 3 reserved */
-        /* 8-255 reseved */
+        /* 8-255 reserved */
       default:
         vinfo->colorimetry.primaries = GST_VIDEO_COLOR_PRIMARIES_UNKNOWN;
         break;
@@ -752,7 +752,7 @@ handle_sequence (GstMpeg2dec * mpeg2dec, const mpeg2_info_t * info)
         /* 0 forbidden */
         /* 2 unspecified */
         /* 3 reserved */
-        /* 8-255 reseved */
+        /* 8-255 reserved */
       default:
         vinfo->colorimetry.matrix = GST_VIDEO_COLOR_MATRIX_UNKNOWN;
         break;
@@ -780,7 +780,7 @@ handle_sequence (GstMpeg2dec * mpeg2dec, const mpeg2_info_t * info)
         /* 0 forbidden */
         /* 2 unspecified */
         /* 3 reserved */
-        /* 9-255 reseved */
+        /* 9-255 reserved */
       default:
         vinfo->colorimetry.transfer = GST_VIDEO_TRANSFER_UNKNOWN;
         break;

ext/x264/gstx264enc.c

@@ -2930,7 +2930,7 @@ plugin_init (GstPlugin * plugin)
 
   GST_INFO ("linked against x264 build: %u", X264_BUILD);
 
-  /* Initialize the static GstX264EncVTable which is overriden in load_x264()
+  /* Initialize the static GstX264EncVTable which is overridden in load_x264()
    * if needed. We can't initialize statically because these values are not
    * constant on Windows. */
   default_vtable.module = NULL;

gst/asfdemux/gstasfdemux.c

@@ -768,7 +768,7 @@ gst_asf_demux_handle_seek_event (GstASFDemux * demux, GstEvent * event)
     /* First try to query our source to see if it can convert for us. This is
        the case when our source is an mms stream, notice that in this case
        gstmms will do a time based seek to get the byte offset, this is not a
-       problem as the seek to this offset needs to happen anway. */
+       problem as the seek to this offset needs to happen anyway. */
     if (gst_pad_peer_query_convert (demux->sinkpad, GST_FORMAT_TIME, seek_time,
             GST_FORMAT_BYTES, &offset)) {
       packet = (offset - demux->data_offset) / demux->packet_size;
@@ -1434,7 +1434,7 @@ gst_asf_demux_get_first_ts (GstASFDemux * demux)
       /* there are some DVR ms files where first packet has TS of 0 (instead of -1) while subsequent packets have
          regular (singificantly larger) timestamps. If we don't deal with it, we may end up with huge gap in timestamps
          which makes playback stuck. The 0 timestamp may also be valid though, if the second packet timestamp continues 
-         from it. I havent found a better way to distinguish between these two, except to set an arbitrary boundary
+         from it. I haven't found a better way to distinguish between these two, except to set an arbitrary boundary
          and disregard the first 0 timestamp if the second timestamp is bigger than the boundary) */
 
       GST_DEBUG_OBJECT (demux,
@@ -2300,7 +2300,7 @@ gst_asf_demux_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
         GstAsfDemuxParsePacketError err;
 
         /* we don't know the length of the stream
-         * check for a chained asf everytime */
+         * check for a chained asf every time */
         if (demux->num_packets == 0) {
           gint result = gst_asf_demux_check_header (demux);
 

gst/dvdlpcmdec/gstdvdlpcmdec.c

@@ -755,7 +755,7 @@ gst_dvdlpcmdec_parse_1394 (GstDvdLpcmDec * dvdlpcmdec, GstAdapter * adapter,
     }
     switch (header & 0x7) {
       case 0x0:                /* 2 channels dual-mono */
-      case 0x1:                /* 2 channles stereo */
+      case 0x1:                /* 2 channels stereo */
         channels = 2;
         break;
       default:

gst/dvdsub/gstdvdsubdec.c

@@ -422,7 +422,7 @@ gst_setup_palette (GstDvdSubDec * dec)
     target2_yuv->A = dec->menu_alpha[i] * 0xff / 0xf;
 
     /* If ARGB flag set, then convert YUV palette to RGB */
-    /* Using integer aritmetic */
+    /* Using integer arithmetic */
     if (dec->use_ARGB) {
       guchar C = target_yuv->Y_R - 16;
       guchar D = target_yuv->U_G - 128;

gst/realmedia/rdtjitterbuffer.c

@@ -144,7 +144,7 @@ rdt_jitter_buffer_reset_skew (RDTJitterBuffer * jbuf)
  *    Cri    : The time of the clock at the receiver for packet i
  *    D + ni : The jitter when receiving packet i
  *
- * We see that the network delay is irrelevant here as we can elliminate D:
+ * We see that the network delay is irrelevant here as we can eliminate D:
  *
  *  recv_diff(i) = (Cri + ni) - (Cr0 + n0))
  *
@@ -420,7 +420,7 @@ duplicate:
  * @jbuf: an #RDTJitterBuffer
  *
  * Pops the oldest buffer from the packet queue of @jbuf. The popped buffer will
- * have its timestamp adjusted with the incomming running_time and the detected
+ * have its timestamp adjusted with the incoming running_time and the detected
  * clock skew.
  *
  * Returns: a #GstBuffer or %NULL when there was no packet in the queue.

gst/realmedia/rtspreal.c

@@ -397,7 +397,7 @@ rtsp_ext_real_parse_sdp (GstRTSPExtension * ext, GstSDPMessage * sdp,
     stream->mime_type = g_strndup (str, stream->mime_type_len);
 
     /* FIXME: Depending on the current bandwidth, we need to select one
-     * bandwith out of a list offered by the server. Someone needs to write
+     * bandwidth out of a list offered by the server. Someone needs to write
      * a parser for strings like
      *
      * #($Bandwidth < 67959),TimestampDelivery=T,DropByN=T,priority=9;

hooks/pre-commit.hook

@@ -1,6 +1,6 @@
 #!/bin/sh
 #
-# Check that the code follows a consistant code style
+# Check that the code follows a consistent code style
 #
 
 # Check for existence of indent, and error out if not present.