docs, gst: typo fixes

https://bugzilla.gnome.org/show_bug.cgi?id=658449

README

@@ -140,7 +140,7 @@ PLATFORMS
 - MacOSX is reported to work; specific audio and video sinks have been written
 - Windows support is experimental but improving.  Output sinks have been
   written but are not yet included in the code. We support
-  - MSys/MingW builds
+  - MSys/MinGW builds
   - Microsoft Visual Studio 6 builds (see win32/README.txt)
 
 INSTALLING FROM PACKAGES

docs/README

@@ -120,7 +120,7 @@ GTK-DOC NOTES
   - add all documented symbols to gstreamer-sections.txt in the proper section
     (default),<SUBSECTION Standard>,<SUBSECTION Private>
   - document at least the Short_Description in tmpl/.sgml
-  - document symbols where they are definied, so that when one changes the
+  - document symbols where they are defined, so that when one changes the
     definition, the chaces are good that docs are updated.
     - document functions, signals in the .c files
     - document structs, typedefs, enums in the .h files

docs/design/draft-buffer2.txt

@@ -5,7 +5,7 @@ This draft document describes a possible design for arbitrary per-buffer
 metadata.
 
 The proposed changes in this document are not ABI/API compatible with the 0.10
-version of GStreamer and should thus only be considered for upcomming unstable
+version of GStreamer and should thus only be considered for upcoming unstable
 versions.
 
 Buffer metadata typically includes properties that give more information about

docs/design/draft-klass.txt

@@ -39,7 +39,7 @@ for wild (application specific) customisation.
 
    Categories are base on _intended usage_ of the element. Some elements
    might have other side-effects (especially for filers/effects). The purpose
-   is to list enough keywords so that applications can do meaningfull filtering,
+   is to list enough keywords so that applications can do meaningful filtering,
    not to completely describe the functionality, that is expressed in caps etc..
 
    * Source        : produces data

docs/design/part-MT-refcounting.txt

@@ -45,7 +45,7 @@ Shared data structures and writability:
   _get_writable() function call. This function will check the refcount of the
   object and if the object is referenced by more than one instance, a copy is
   made of the object that is then by definition only referenced from the calling
-  thread. This new copy is then modifyable without being visible to other
+  thread. This new copy is then modifiable without being visible to other
   refcount holders.
 
   This technique is used for information objects that, once created, never

docs/design/part-TODO.txt

@@ -27,7 +27,7 @@ API/ABI
 - Optimize negotiation. We currently do a get_caps() call when we link pads,
   which could potentially generate a huge list of caps and all their
   combinations, we need to avoid generating these huge lists by generating them
-  incrementaly when needed. We can do this with a gst_pad_iterate_caps() call.
+  incrementally when needed. We can do this with a gst_pad_iterate_caps() call.
   We also need to incrementally return intersections etc, for this.
 
 - Elements in a bin have no clue about the final state of the parent element

docs/design/part-activation.txt

@@ -63,7 +63,7 @@ fakesrc:src's activation function is then called.
 
 Note that it does not make sense to set an activation function on a
 source pad. The peer of a source pad is downstream, meaning it should
-have been activated first. If it was activated in PULL mode, the the
+have been activated first. If it was activated in PULL mode, the
 source pad should have already had activate_pull() called on it, and
 thus needs no further activation. Otherwise it should be in PUSH mode,
 which is the choice of the default activation function.

docs/design/part-block.txt

@@ -163,7 +163,7 @@ Issues
 
 When an EOS event has passed a pad and the pad is set to blocked, the block will
 never happen because no data is going to flow anymore. One possibility is to
-keep track of the pad's EOS state and make the block succeed immediatly. This is
+keep track of the pad's EOS state and make the block succeed immediately. This is
 not yet implemenented.
 
 When dynamically reconnecting pads, some events (like NEWSEGMENT, EOS,

docs/design/part-buffering.txt

@@ -11,7 +11,7 @@ live sources.
 
 We want to be able to implement the following features:
 
- - buffering up to a specifc amount of data, in memory, before starting playback
+ - buffering up to a specific amount of data, in memory, before starting playback
    so that network fluctuations are minimized.
  - download of the network file to a local disk with fast seeking in the
    downloaded data. This is similar to the quicktime/youtube players.
@@ -79,7 +79,7 @@ Some use cases:
 
   The application can use the BUFFERING query to get the estimated download time
   and match this time to the current/remaining playback time to control when
-  playback should start to have a non-interupted playback experience.
+  playback should start to have a non-interrupted playback experience.
 
 
  * Timeshifting

docs/design/part-bufferlist.txt

@@ -81,7 +81,7 @@ group.
 Metadata
 ~~~~~~~~
 
-Each of the buffers inside the bufferlist can have metadata assiociated with it.
+Each of the buffers inside the bufferlist can have metadata associated with it.
 
 The metadata of the bufferlist is always the metadata of the first buffer of the
 first group in the bufferlist. This means that:

docs/design/part-caps.txt

@@ -15,6 +15,6 @@ produce (see part-pads.txt and part-negotiation.txt).
 Caps are also attached to buffers to describe to content of the data
 pointed to be the buffer.
 
-Various methods exist to work with the media types such as substracting
+Various methods exist to work with the media types such as subtracting
 or intersecting.
 

docs/design/part-clocks.txt

@@ -3,7 +3,7 @@ Clocks
 
 The GstClock returns a monotonically increasing time with the method 
 _get_time(). Its accuracy and base time depends on the specific clock 
-implementation but time is always expessed in nanoseconds. Since the
+implementation but time is always expressed in nanoseconds. Since the
 baseline of the clock is undefined, the clock time returned is not
 meaningful in itself, what matters are the deltas between two clock
 times.

docs/design/part-element-sink.txt

@@ -179,7 +179,7 @@ sink overview
       STREAM_UNLOCK
       break
     NEWSEGMENT:
-      # the newsegment must be used to clip incomming
+      # the newsegment must be used to clip incoming
       # buffers. Then then go into the queue as non-prerollable
       # items used for syncing the buffers
       STREAM_LOCK

docs/design/part-element-transform.txt

@@ -25,7 +25,7 @@ the following things:
 Some transform elements can operate in different modes:
 
  - passthrough (no changes are done on the input buffers)
- - in-place (changes made directly to the incomming buffers without requiring a
+ - in-place (changes made directly to the incoming buffers without requiring a
    copy or new buffer allocation)
  - metadata changes only 
  
@@ -266,7 +266,7 @@ state. We can identify these steady states:
 	input buffer is transformed into the output buffer. The flow is exactly
 	the same as the case with the same-caps negotiation. (DCC)
  
-We can immeditatly observe that the copy transform states will need to
+We can immediately observe that the copy transform states will need to
 allocate a buffer from a downstream element using pad-alloc. When the transform
 element is receiving a non-writable buffer in the in-place state, it will also
 need to perform a pad-alloc. There is no reason why the passthrough state would
@@ -300,7 +300,7 @@ generated in the chain function is handled above in the copy-transform and the
 in-place transform when the input buffer is not writable or the input buffer
 size is smaller than the output size.
 
-We are left with the last case (proxy an incomming pad-alloc or not). We have 2
+We are left with the last case (proxy an incoming pad-alloc or not). We have 2
 possibilities here:
 
    - pad-alloc is called with the same caps as are currently being handled by
@@ -413,7 +413,7 @@ retrieve the size. There are two functions:
     
     Given a caps and a size on one pad, and a caps on the other pad, calculate
     the size of the other buffer. This function is able to perform all size
-    transforms and is the prefered method of transforming a size.
+    transforms and is the preferred method of transforming a size.
 
   - get_unit_size()
 

docs/design/part-events.txt

@@ -121,11 +121,11 @@ Before sending buffers, an element must send a NEWSEGMENT event. An element is
 free to refuse buffers if they were not preceeded by a NEWSEGMENT event.
 
 Elements that sync to the clock should store the NEWSEGMENT start and end values
-and substract the start value from the buffer timestamp before comparing
+and subtract the start value from the buffer timestamp before comparing
 it against the stream time (see part-clocks.txt).
 
 An element is allowed to send out buffers with the NEWSEGMENT start time already
-substracted from the timestamp. If it does so, it needs to send a corrected
+subtracted from the timestamp. If it does so, it needs to send a corrected
 NEWSEGMENT downstream, ie, one with start time 0.
 
 A NEWSEGMENT event should be generated as soon as possible in the pipeline and

docs/design/part-framestep.txt

@@ -113,7 +113,7 @@ The step event is created with the following fields in the structure:
      direction.
 
   "flush", G_TYPE_BOOLEAN
-     when flushing is TRUE, the step is performed immediatly: 
+     when flushing is TRUE, the step is performed immediately:
 
       - In the PAUSED state the pipeline loses the PAUSED state, the requested
         amount of data is skipped and the pipeline prerolls again when a
@@ -140,7 +140,7 @@ The step event is created with the following fields in the structure:
      Signal that this step operation is an intermediate step, part of a series
      of step operations. It is mostly interesting for stepping in the PAUSED state
      because the sink will only perform a preroll after a non-intermediate step
-     operation completes. Intermediate steps are usefull to flush out data from
+     operation completes. Intermediate steps are useful to flush out data from
      other sinks in order to not cause excessive queueing. In the PLAYING state
      the intermediate flag has no visual effect. In all states, the intermediate
      flag is passed to the corresponding GST_MESSAGE_STEP_DONE.

docs/design/part-gstelement.txt

@@ -51,7 +51,7 @@ gst_element_add_pad(element,pads):
 	Fails if either the element or pad are either NULL or not what they
 	claim to be.  Should fail if the pad already has a parent. Should fail
 	if the pad is already owned by the element.  Should fail if there's
-	already a pad by that name in the the list of pads.
+	already a pad by that name in the list of pads.
 
 pad = gst_element_get_pad(element,"padname"):
 	Searches through the list of pads 

docs/design/part-gstghostpad.txt

@@ -406,7 +406,7 @@ are already active, so it ignores them.
 Similarly in (4), activating D will cause the activation of all of the
 rest of the pads, in this order: C d c b a B A. Then when the state
 change gets to the other elements they are already active, and in fact
-data flow is already occuring.
+data flow is already occurring.
 
 So, from these scenarios, we can distill how ghost pad activation
 functions should work:

docs/design/part-latency.txt

@@ -95,7 +95,7 @@ capture pipelines.
   * the sink will receive the buffer with timestamp 0 at time >= D. At this
     point the buffer is too late already and might be dropped. This state of
     constantly dropping data will not change unless a constant latency
-    correction is added to the incomming buffer timestamps.
+    correction is added to the incoming buffer timestamps.
 
   The problem is due to the fact that the sink is set to (pending) PLAYING
   without being prerolled, which only happens in live pipelines. 

docs/design/part-messages.txt

@@ -103,7 +103,7 @@ GST_MESSAGE_ELEMENT:
 GST_MESSAGE_SEGMENT_START:
 
   An element started playback of a new segment. This message is not forwarded
-  the the application but is used internally to schedule SEGMENT_DONE messages.
+  the application but is used internally to schedule SEGMENT_DONE messages.
 
 GST_MESSAGE_SEGMENT_DONE:
 

docs/design/part-missing-plugins.txt

@@ -40,7 +40,7 @@ A three step process:
   There is not much useful information we can give about how to resolve this 
   issue. It is possible to use the first N bytes of the data to determine the
   type (and needed plugin) on the server. We don't explore this option in this
-  document yet, but the proposal is flexible enough to accomodate this in the
+  document yet, but the proposal is flexible enough to accommodate this in the
   future should the need arise.
 
  - missing demuxer

docs/design/part-negotiation.txt

@@ -42,12 +42,12 @@ The basics of negotiation are as follows:
    The core will automatically call the set_caps function for this purpose
    when it is installed on the sink or source pad.
 
- - When requesting a buffer from a bufferpool, the prefered type should
+ - When requesting a buffer from a bufferpool, the preferred type should
    be passed to the buffer allocation function. After receiving a buffer
    from a bufferpool, the datatype should be checked again.
 
  - A bufferpool allocation function should try to allocate a buffer of the
-   prefered type. If there is a good reason to choose another type, the
+   preferred type. If there is a good reason to choose another type, the
    alloc function should see if that other type is accepted by the other
    element, then allocate a buffer of that type and attach the type to the
    buffer before returning it.

docs/design/part-qos.txt

@@ -9,7 +9,7 @@ clock and typically happens in the sinks when they synchronize buffers
 against the clock.
 
 The measurements result in QOS events that aim to adjust the datarate
-in one or more upstream elements. Two types of adjustements can be
+in one or more upstream elements. Two types of adjustments can be
 made:
 
   - short time "emergency" corrections based on latest observation
@@ -350,7 +350,7 @@ in audio are more disturbing than dropping video frames. Also video requires in
 general more processing than audio.
 
 Normally there is a threshold for when buffers get dropped in a video sink. Frames
-that arrive 20 milliseconds late are still rendered as it is not noticable for
+that arrive 20 milliseconds late are still rendered as it is not noticeable for
 the human eye.
 
 A QoS message is posted whenever a (part of a) buffer is dropped.

docs/design/part-scheduling.txt

@@ -8,7 +8,7 @@ Pushing
 ~~~~~~~
 
 A pad can produce data and push it to the next pad. A pad that behaves this way
-exposes a loop function that will be called repeadedly until it returns false. 
+exposes a loop function that will be called repeatedly until it returns false.
 The loop function is allowed to block whenever it wants. When the pad is deactivated
 the loop function should unblock though.
 
@@ -24,7 +24,7 @@ Pulling
 
 Pads that operate in pulling mode can only pull data from a pad that exposes the
 pull_range function. In this case, the sink pad exposes a loop function that will be
-called repeadedly until the task is stopped.
+called repeatedly until the task is stopped.
 
 After pulling data from the peer pad, the loop function will typically call the
 push function to push the result to the peer sinkpad.

docs/design/part-seeking.txt

@@ -178,12 +178,12 @@ Summary:
 
  - if the KEY_UNIT flag is *not* specified, the demuxer/parser should
    start pushing data from a key unit preceding the seek position
-   (or from the the seek position if that falls on a key unit), and
+   (or from the seek position if that falls on a key unit), and
    the start of the new segment should be the requested seek position.
 
  - if the KEY_UNIT flag is specified, the demuxer/parser should start
    pushing data from the key unit nearest the seek position (or from
-   the the seek position if that falls on a key unit), and
+   the seek position if that falls on a key unit), and
    the start of the new segment should be adjusted to the position of
    that key unit which was nearest the requested seek position (ie.
    the new segment start should be the position from which data is

docs/design/part-states.txt

@@ -179,7 +179,7 @@ function returns a GstElementStateReturn.
 
    * If the element aborts the ASYNC state change due to an error within the 
      specified timeout, this function returns FAILURE with the state set to last
-     successfull state and pending set to the last attempt. The element should 
+     successful state and pending set to the last attempt. The element should
      also post an error message on the bus with more information about the problem.
 
 
@@ -201,7 +201,7 @@ error.
 If all the children return SUCCESS, the function returns SUCCESS as well. 
 
 If one of the children returns FAILURE, the function returns FAILURE as well. In
-this state it is possible that some elements successfuly changed state. The 
+this state it is possible that some elements successfully changed state. The
 application can check which elements have a changed state, which were in error
 and which were not affected by iterating the elements and calling _get_state()
 on the elements.

docs/design/part-stream-status.txt

@@ -9,7 +9,7 @@ thread changes. The purpose of this message is to allow the application to
 interact with the streaming thread properties, such as the thread priority or
 the threadpool to use.
 
-We accomodate for the following requirements:
+We accommodate for the following requirements:
 
  - Application is informed when a streaming thread is about to be created. It
    should be possible for the application to suggest a custom GstTask.

docs/faq/Makefile.am

@@ -19,7 +19,7 @@ FIG_SRC = $(notdir $(wildcard $(srcdir)/*.fig))
 # extra sources to copy in build directory
 EXTRA_SRC = 
 
-### this is the generic bit and you shouln't need to change this
+### this is the generic bit and you shouldn't need to change this
 
 # get the generic docbuilding Makefile stuff
 include $(srcdir)/../manuals.mak

docs/faq/dependencies.xml

@@ -57,7 +57,7 @@ Does GStreamer offer support for DVD decoder cards like dxr2/3 ?
       <answer>
         <para>
 We do have support for the dxr3, although dxr2 support is unknown. 
-GStreamer can easily accomodate hardware acceleration by writing new 
+GStreamer can easily accommodate hardware acceleration by writing new
 device-specific elements.
         </para>
       </answer>

docs/faq/general.xml

@@ -80,7 +80,7 @@ application doesn't play back your files, you can help us solve that problem
 by <ulink url="http://bugzilla.gnome.org">filing an enhancement request
 bug</ulink> for that format. If you have it, please provide: 
 <itemizedlist>
-  <listitem><para>links to other players, preferrably Open Source and working 
+  <listitem><para>links to other players, preferably Open Source and working
   on Unix</para></listitem>
   <listitem><para>links to explanations of the format.</para></listitem>
   <listitem><para>ways to obtain mediafiles in that format to test.

docs/manual/Makefile.am

@@ -22,7 +22,7 @@ FIG_SRC = $(notdir $(wildcard $(srcdir)/*.fig))
 # extra sources to copy in build directory
 EXTRA_SRC = 
 
-### this is the generic bit and you shouln't need to change this
+### this is the generic bit and you shouldn't need to change this
 
 # get the generic docbuilding Makefile stuff
 include $(srcdir)/../manuals.mak

docs/manual/advanced-clocks.xml

@@ -53,7 +53,7 @@
       <function>gst_clock_get_time ()</function>. The clock-time does not
       need to start at 0. The pipeline, which contains the global clock that
       all elements in the pipeline will use, in addition has a <quote>base
-      time</quote>, which is the clock time at the the point where the
+      time</quote>, which is the clock time at the point where the
       pipeline went to the PLAYING state. Each element can subtract the
       <quote>base time</quote> from the clock-time to know the current
       running time.

docs/manual/advanced-dparams.xml

@@ -7,7 +7,7 @@
       The controller subsystem offers a lightweight way to adjust gobject
       properties over stream-time. Normaly these properties are changed using
       <function>g_object_set()</function>. Timing those calls reliably so that
-      the changes affect certain stream times is close to imposible. The
+      the changes affect certain stream times is close to impossible. The
       controller takes time into account. It works by attaching control-sources
       to properties. Control-sources can provide new values for the properties
       for a given timestamp. At run-time the elements continously pull values
@@ -30,7 +30,7 @@
     </para>
     <para>
       The <filename>gstreamer-controller</filename> library needs to be initialized
-      when your application is run.  This can be done after the the GStreamer
+      when your application is run.  This can be done after the GStreamer
       library has been initialized.
     </para>
     <programlisting>

docs/manual/basics-elements.xml

@@ -525,7 +525,7 @@ main (int   argc,
           video sinks can already play the first frame (since this does
           not affect the clock yet). Autopluggers could use this same
           state transition to already plug together a pipeline. Most other
-          elements, such as codecs or filters, do not need to explicitely
+          elements, such as codecs or filters, do not need to explicitly
           do anything in this state, however.
         </para>
       </listitem>

docs/manual/basics-init.xml

@@ -3,7 +3,7 @@
   <para> 
     When writing a &GStreamer; application, you can simply include
     <filename>gst/gst.h</filename> to get access to the library
-    functions. Besides that, you will also need to intialize the
+    functions. Besides that, you will also need to initialize the
     &GStreamer; library.
   </para>
 

docs/manual/basics-pads.xml

@@ -37,7 +37,7 @@
       request. The meaning of those three types is exactly as it says:
       always pads always exist, sometimes pad exist only in certain
       cases (and can disappear randomly), and on-request pads appear
-      only if explicitely requested by applications.
+      only if explicitly requested by applications.
     </para>
 
     <sect2 id="section-pads-dynamic">

docs/manual/diagrams-general.svg

@@ -12581,7 +12581,7 @@
          style="font-size:12px"
          y="-434.5477"
          x="554.7406"
-         sodipodi:role="line">- effetcs</tspan><tspan
+         sodipodi:role="line">- effects</tspan><tspan
          id="tspan4637"
          style="font-size:12px"
          y="-419.5477"
@@ -14283,7 +14283,7 @@
          style="font-size:12px"
          y="-1005.6619"
          x="499.47424"
-         sodipodi:role="line">- effetcs</tspan><tspan
+         sodipodi:role="line">- effects</tspan><tspan
          id="tspan6117"
          style="font-size:12px"
          y="-990.66187"

docs/manual/highlevel-components.xml

@@ -163,7 +163,7 @@ main (gint   argc,
         <para>
           Supports stream selection and disabling. If your media has
           multiple audio or subtitle tracks, you can dynamically choose
-          which one to play back, or decide to turn it off alltogther
+          which one to play back, or decide to turn it off altogether
           (which is especially useful to turn off subtitles). For each
           of those, use the <quote>current-text</quote> and other related
           properties.

docs/manual/intro-gstreamer.xml

@@ -23,7 +23,7 @@
     </para> 
 
     <para>
-      One of the the most obvious uses of &GStreamer; is using it to build
+      One of the most obvious uses of &GStreamer; is using it to build
       a media player. &GStreamer; already includes components for building a
       media player that can support a very wide variety of formats, including
       MP3, Ogg/Vorbis, MPEG-1/2, AVI, Quicktime, mod, and more. &GStreamer;,

docs/pwg/Makefile.am

@@ -19,7 +19,7 @@ FIG_SRC = $(notdir $(wildcard $(srcdir)/*.fig))
 # extra sources to copy in build directory
 EXTRA_SRC = 
 
-### this is the generic bit and you shouln't need to change this
+### this is the generic bit and you shouldn't need to change this
 
 # get the generic docbuilding Makefile stuff
 include $(srcdir)/../manuals.mak

docs/pwg/advanced-tagging.xml

@@ -81,7 +81,7 @@
       element, it might be a good idea to add it to <filename>gsttag.c</filename>
       instead. That's up to you to decide. If you want to do it in your own
       element, it's easiest to register the tag in one of your class init
-      functions, preferrably <function>_class_init ()</function>.
+      functions, preferably <function>_class_init ()</function>.
     </para>
     <programlisting>
 <![CDATA[

docs/pwg/intro-basics.xml

@@ -272,7 +272,7 @@
       <title>The Basic Types</title>
       <para>
         &GStreamer; already supports many basic media types. Following is a
-        table of a few of the the basic types used for buffers in
+        table of a few of the basic types used for buffers in
         &GStreamer;. The table contains the name ("mime type") and a
         description of the type, the properties associated with the type, and
         the meaning of each property. A full list of supported types is

docs/pwg/intro-preface.xml

@@ -27,7 +27,7 @@
     </para> 
 
     <para>
-      One of the the most obvious uses of &GStreamer; is using it to build
+      One of the most obvious uses of &GStreamer; is using it to build
       a media player. &GStreamer; already includes components for building a
       media player that can support a very wide variety of formats, including
       MP3, Ogg/Vorbis, MPEG-1/2, AVI, Quicktime, mod, and more. &GStreamer;,

docs/pwg/other-base.xml

@@ -151,7 +151,7 @@ gst_my_sink_class_init (GstMySinkClass * klass)
         <listitem>
           <para>
             Features can be added to all audiosinks by making a change in the
-            base class, which makes maintainance easy.
+            base class, which makes maintenance easy.
           </para>
         </listitem>
         <listitem>
@@ -192,7 +192,7 @@ gst_my_sink_class_init (GstMySinkClass * klass)
           <para>
             By adding new features to <classname>GstVideoSink</classname>, it
             will be possible to add extensions to videosinks that affect all of
-            them, but only need to be coded once, which is a huge maintainance
+            them, but only need to be coded once, which is a huge maintenance
             benefit.
           </para>
         </listitem>

docs/pwg/other-source.xml

@@ -58,7 +58,7 @@ gst_my_source_get (GstPad *pad)
       Those will continuously describe the current state of the stream.
       Query functions can be used to get stream properties such as current
       position and length. This can be used by fellow elements to convert
-      this same value into a different unit, or by appliations to provide
+      this same value into a different unit, or by applications to provide
       information about the length/position of the stream to the user.
       Conversion functions are used to convert such values from one unit
       to another. Lastly, events are mostly used to seek to positions

docs/random/autoplug2

@@ -43,7 +43,7 @@ provided by various plugins.
 
 We will create a little pipeline to detect the media type by connecting
 a disksrc element to a typefind element. The typefind element will 
-repeadedly call all registered typefind functions with the buffer it 
+repeatedly call all registered typefind functions with the buffer it
 receives on its sink pad. when a typefind function returns a non NULL
 GstCaps*, that caps is set to the sink pad of the typefind element and
 a signal is emitted to notify the app.
@@ -195,7 +195,7 @@ for the second sink the following is needed:
 
 Note that for the audio connection the element list "mpeg1parse, mp3parse,
 mpg123" would also connect the srccaps to the audiosink caps. Since the
-"mpeg1parse, mad" list is shorter, it it always prefered by the autoplugger.
+"mpeg1parse, mad" list is shorter, it it always preferred by the autoplugger.
 
 We now have two lists of elementfactories.
 

docs/random/bbb/optional-properties

@@ -54,7 +54,7 @@ aligned video) and pixel-aspect-ratio (default value being 1/1). Adding p-a-r
 was in this case an example of 2bII, whereas stride is 2bI. The reason for
 this is simple: adding pixel-aspect-ratio to some element and not to others
 could lead to misunderstanding size. However, this is not a regression,
-because not adding it alltogether would lead to the same misunderstanding.
+because not adding it altogether would lead to the same misunderstanding.
 In both cases, the result would be wrongly sized video. Therefore, there
 is no regression and there is a bugfix, so this is fine. Obviously, the
 optional property is in this case very specifically a temporary solution.

docs/random/bbb/streamselection

@@ -4,13 +4,13 @@ Stream selection
 1. Problem
 URIs (that being either a media stream or a media stream plus subtitle) can
 contain multiple streams of a type (audio, subtitle). A user has to be given
-the option of selecting one of those streams (or none alltogether).
+the option of selecting one of those streams (or none altogether).
 
 2. Implementation ideas
 Stream selection, in GStreamer, has to be integrated at the player plugging
 level, which is (in the case of Totem) playbin. Playbin offers a feature to
 'mute' a stream (which means that no processing is done on that stream
-alltogether, saving the decoding step). Currently, playbin will select the
+altogether, saving the decoding step). Currently, playbin will select the
 first occurrence of a stream type and mute all others. A queue is connected
 (for pre-roll) to the active stream. What is missing here is a way to change
 the active stream.
@@ -30,7 +30,7 @@ stream was linked. Alternatively, a switch-like element is used, which
 requires no replugging. Pad disabling/enabling is then enough. This also
 makes relinking less painful. The switch-like element needs to proxy the
 active pads' caps. However, since those caps are (in practice) always the
-same accross streams, caps setting will (inside the core) immediately
+same across streams, caps setting will (inside the core) immediately
 return success.
 The switch-like element simply works like this:
 =

docs/random/caps

@@ -162,7 +162,7 @@ As with option2, a global plan will be built. At runtime the src pads
 will actually specify the capabilities they need for any element that
 wants to be connected to its source pads.
 
-In this case we specifiy the capabilities for all the sink pads of an
+In this case we specify the capabilities for all the sink pads of an
 element at create time. The capabilities of the src pads would only
 become available when data has been processed by the element.
 

docs/random/company/gvadec.txt

@@ -209,7 +209,7 @@ for themselves - most of the time depending on caps on other pads in the
 element. The core then takes those, intersects them and if the intersection
 isn't empty fixates them. Fixation is a process that selects the best possible
 fixed caps from a caps. A fixed caps is a caps that describes only one format
-and cannot be reduced further. After both pads acdepted the fixed caps, its
+and cannot be reduced further. After both pads accepted the fixed caps, its
 format is then used to describe the contents of the buffers that are passed on
 this link. Caps can be serialized and deserialized to a string representation.
 [Add: a medium complex caps description (audioconvert?)]

docs/random/ensonic/draft-bufferpools.txt

@@ -39,7 +39,7 @@ What is needed
 
 Elements that sink raw data buffers of usualy constant size would like to 
 maintain a bufferpool. These could be sinks or encoders. We need mechanims to
-select and dynamicaly update:
+select and dynamically update:
 
   - the bufferpool owners in a pipeline
   - the bufferpool sizes

docs/random/ensonic/embedded.txt

@@ -21,7 +21,7 @@ encapsulated in the indexer strategy.
 Autopluggers like playbin and decodebin use the element caps plus static ranks
 to create piplines.
 The rank of an elemnt right now refers to the quality/maturity of the element.
-Elements with higher rank should be functionaly more complete. If we have
+Elements with higher rank should be functionally more complete. If we have
 multiple elements that are feature complete there is a draw.
 
 There are more decission criteria thinkable:

docs/random/ensonic/media-device-daemon.txt

@@ -38,5 +38,5 @@ components
       channelstrip
   - state-control (play, pause/mute)
     - it would be useful if one app could pause/mute others
-    - think of a voip-client, if there is an incomming call, if pauses your
+    - think of a voip-client, if there is an incoming call, if pauses your
       media-player, or mutes the monitoring of your recording app

docs/random/ensonic/plugindocs.txt

@@ -33,7 +33,7 @@ Details
 ------------------
 - read data from inspect/plugin-<pluginname>.xml
 - insert/overwrite "Short Description" and "Long Description" in tmpl/
-- the "Long Description" contains a <xi:inlcude> for xml/element-<name>-details.xml
+- the "Long Description" contains a <xi:include> for xml/element-<name>-details.xml
 
 3.) common/plugins.xsl
 ----------------------

docs/random/ensonic/profiling.txt

@@ -89,7 +89,7 @@ $Id$
 
 = PerformanceMonitor =
 Write a ld-preload lib that can gather data from gstreamer and logs it to files.
-The idea is not avoid adding API for performance meassurement to gstreamer.
+The idea is not avoid adding API for performance measurement to gstreamer.
 
 == Services ==
 library provides some common services used by the sensor modules.
@@ -97,7 +97,7 @@ library provides some common services used by the sensor modules.
 * timestamps
 
 == Sensors ==
-Sensors do meassurements and deliver timestampe performance data.
+Sensors do measurements and deliver timestampe performance data.
 * bitrates and latency via gst_pad_push/pull per link
 * qos ratio via gst_event_new_qos(), gst_pad_send_event()
 * cpu/mem via get_rusage
@@ -125,7 +125,7 @@ timestamp [qos-ratio] [cpu-load={sum,17284,17285}]
 ** should we have the log config in the header or in some separate config?
    - if config, we just specify the config when capturing put that
      in the first log line
-   - otheriwse the analyzer ui has to parse it from the first line
+   - otherwise the analyzer ui has to parse it from the first line
 
 == Running ==
 LD_PRELOAD=libgstperfmon.so GST_PERFMON_DETAILS="qos-ratio,cpu-load=all" <application>

docs/random/eos

@@ -82,12 +82,12 @@ case 3)
   the eos handler returns false because both queues return false on the
   eos request. the parent removes fakesrc as an EOS provider.
 
-  queue1 and queue2 were responisble for the EOS delay and so they get
+  queue1 and queue2 were responsible for the EOS delay and so they get
   added to the bin as possible EOS providers.
 
   after the queues have sent out their last buffer, they calls eos on their
   src pads.
-  the parent allready has the two queues in the EOS provider list so they dont
+  the parent already has the two queues in the EOS provider list so they dont
   get added twice.
   the two queues perform gst_pad_eos () on their pads when the queue is empty,
   the parent removes the EOS providers from its list, when the list is empty,
@@ -122,12 +122,12 @@ case 4)
   the eos handler returns false because queue2 returns false on the
   eos request. the parent removes fakesrc as an EOS provider.
 
-  queue2 was responisble for the EOS delay and so it gets added to the bin 
+  queue2 was responsible for the EOS delay and so it gets added to the bin
   as a possible EOS provider.
 
   after the queue2 has sent its last buffer, it performs gst_pad_eos on its
   src pad.
-  the parent allready has the queue2 in the list of EOS providers so it does not
+  the parent already has the queue2 in the list of EOS providers so it does not
   get added twice.
   queue2 finally fires the EOS signal and the parent removes the EOS provider 
   from its list, when the list is empty, the parent fires EOS.

docs/random/hierarchy

@@ -12,7 +12,7 @@ better layout for it now. Some things to consider:
  - The plugins can be grouped together by the media type they
    operate on or by the way they work (decoder/encoder)
    
-In GStreamer all plugins are techically filters, the only way they
+In GStreamer all plugins are technically filters, the only way they
 can be considered sources or sinks (input/output) elements is
 by the absence of src/sink pads. At first sight the source/filter/
 sink distinction is quite useless because most of the plugins

docs/random/i18n

@@ -10,7 +10,7 @@ Internationalization notes
   - should only use bindtextdomain to tie a domain to a locale dir
   - use dgettext (possibly disguised as _) to translate from a set domain
 
-- How to make your plug-in code translateable:
+- How to make your plug-in code translatable:
   - include <gst/gst-i18n-plugin.h> in all files that mark strings for
     translation, or do the bindtextdomain call
   - in plugin_init, add a block like this:

docs/random/interfaces

@@ -311,7 +311,7 @@ That's all! It would look similar for FB & co.
 4c) user input
 --------------
 And yes, user input could be an interface too. Even better, it
-should definately be. And wasn't this one of our key issues for
+should definitely be. And wasn't this one of our key issues for
 0.8.0?
 
 No code here. Go implement it, lazy ass!

docs/random/negotiation

@@ -155,7 +155,7 @@ Notes for converters:
      structure, perform whatever other setup is necessary for your
      element, and return GST_PAD_LINK_OK.
 
-   - Otherwise, you're not not using passthrough, and may need to
+   - Otherwise, you're not using passthrough, and may need to
      change the caps on the otherpad to match the given format.
 
    - If the otherpad is not negotiated (!gst_pad_is_negotiated()),

docs/random/omega/sched/chains

@@ -71,7 +71,7 @@ both entries is hard, because the idea at this point is that there's only a sing
 Does this mean that we should make mix a DECOUPLED element?  That would fix it to some extent, giving us three chains 
 in the above case.  Each eq chain would be driven by the eq element, pulling from the queue and pushing into the mixer.  
 The mixer -> queue chain is problematic, because there is no possibly entry.  The mixer side has no _get function 
-(since the push always happens upon the receipt of a buffer from the second sink pad), which means that that those two 
+(since the push always happens upon the receipt of a buffer from the second sink pad), which means that those two
 pads have no possible entrance.
 
 Cothreads make this case much easier, since the mix element would drive things, forcing the eq elements to pull and 

docs/random/omega/testing/framework

@@ -26,7 +26,7 @@ preconditions:
 	nothing
 action:
 	curparent = gst_element_get_parent(object);
-validaton:
+validation:
 	curparent == object->parent
 	curparent == parent
 cleanup:

docs/random/plugins

@@ -65,7 +65,7 @@ We will have the following methods for (implicitly) loading plugins:
 
  gst_plugin_load (name)
 
-  The plugin will be located in the tree and if it allready loaded, the
+  The plugin will be located in the tree and if it already loaded, the
   function returns. If it is not loaded, the XML entry is removed and
   the plugin is loaded. The plugin_init is called so that the XML tree
   is reconstructed.
@@ -73,7 +73,7 @@ We will have the following methods for (implicitly) loading plugins:
  gst_elementfactory_create (factory, name);
 
   The plugin providing the element will be located, if the plugin is
-  allready loaded, an element with the given name is created.
+  already loaded, an element with the given name is created.
   if the plugin is not loaded, gst_plugin_load is called.
   the loaded factory is located again and the element is created.
 

docs/random/rtp

@@ -5,7 +5,7 @@ A brief description of the RTP protocol.
 
 The RTP protocol uses two connections: one for passing data and another for control. The control connection is used for starting, finishing, and passing statistics of lost packets. On the data connection, packets are transmitted containing the media. These packets have a 7 bit field that says what codec is the data transmitted with. This codec is variable. However data cannot change from audio to video in packets of the same connection. All the packets belong to the same logical stream, that is, for instanace, to the same speech, or to the same song. Different codecs in the same stream is used, for instance, to insert comfort noise.
 
-Each codec is packeted in a specific way in RTP packets. This is necessary to minimize the damage produced by lost packets. Therefore, packets should be arranged so that each packet can be decoded independently. If a packet is lost, that shoudn't preclude the decoding of following packets.
+Each codec is packeted in a specific way in RTP packets. This is necessary to minimize the damage produced by lost packets. Therefore, packets should be arranged so that each packet can be decoded independently. If a packet is lost, that shouldn't preclude the decoding of following packets.
 
 
 Suggested implementation:
@@ -38,7 +38,7 @@ Ronald suggested to use inheritance. Thus the user should insert a codec specifi
 
 udpsrc ! rtp-mp3 ! mp3decode ! osssink
 
-Reuse of RTP logic would be achived through inhertiance. 
+Reuse of RTP logic would be achieved through inheritance.
 
 This looks more logical, because inheritance reflects the fact that rtp-mp3 "is an" specialization of rtp. However, there are several issues. As stated above, it is posible in RTP to switch the encoding of the media at any time. If this happens, some state must be kept, such as statistics of packets received and sent. 
 

docs/random/slomo/controller.txt

@@ -6,7 +6,7 @@ Implementation:
 
 Ideas and plans:
   - Deprecate control-rate property and add a control-period property
-    that does the same and is named appropiately. Damn confusing names.
+    that does the same and is named appropriately. Damn confusing names.
 
   - gst_object_suggest_next_sync() will not be used at all anymore.
     Note this in the docs and explain correct usage in elements.

docs/random/sources

@@ -266,7 +266,7 @@ Chained:
        -
 
 As usual, is likely to be an entry into a Bin.  The Bin iterate code must
-explicitely pull a buffer and pass it on to the peer. 
+explicitly pull a buffer and pass it on to the peer.
 
 Cothreaded: 
 

docs/random/streamheader

@@ -2,7 +2,7 @@ A lot of data streams need a few initial buffers to make sense of the
 data stream.  With these initial buffers, it can pick up at any point in
 the rest of the data stream.
 
-Exampes:
+Examples:
 - Vorbis and Theora have three initial Ogg packets
 - MPEG4 has codec data
 - GDP protocol serializes the initial new_segment event and the initial caps

docs/random/testing/syntax

@@ -22,7 +22,7 @@ tcS: id2, element, signalname
 ...
 tcI: the number of iterations on the top bin 
 tcT: a timeout value in mSecs
-tcR: id1,1,id2,1,.. (the pattern of signals trigered)
+tcR: id1,1,id2,1,.. (the pattern of signals triggered)
 or 
 tcR: id1==id2,... (denote an equal number of signals)
 /n

docs/random/types2

@@ -97,7 +97,7 @@ the equivalence of both types, even it they have a different mime type.
 5. type hierarchy
 -----------------
 
-some plugins can ouput a specific subset of an allready existing type.
+some plugins can ouput a specific subset of an already existing type.
 
 example:
 

docs/random/uraeus/gstreamer_and_midi.txt

@@ -234,7 +234,7 @@ otherwise it can stay in individial projects.
 On using dparams for MIDI
 -------------------------
 
-You might might want to look into using dparams if:
+You might want to look into using dparams if:
 
 - you wanted your control parameters to change at a higher rate thanyour buffer
   rate (think zipper noise and sample-granularity-interpolation)

docs/random/vis-transform

@@ -26,7 +26,7 @@ unless they're 1/2^n, and even then it's significantly heavier (you'd have
 to mask the top n bits of each color out).
 
 This SCREAMS for MMX, in case you haven't figured it out yet.
-Unfortunatley, MMX is only directly useful for the scalar matrix, unless
+Unfortunately, MMX is only directly useful for the scalar matrix, unless
 you do a trick where all the pixels in that fit in 64 bits (8 8bit, 4
 16bit, or 2 32bit) are always moved in a group.  This is very possible,
 and might be a significant perf increase by being able to use MMX all the

docs/random/wtay/caps-negociation

@@ -193,7 +193,7 @@ of the pads negotiation functions returns the caps unmodified.
 
 The element can also return a NULL pointer if it has run out of
 options for the caps structure. When this happens, both pads are set
-the the NULL caps again and the pad connnection is broken.
+the NULL caps again and the pad connnection is broken.
 
 The negotiation process is stopped after a fixed number of tries,
 when the counter has reached some limit. This limit is typically

docs/random/wtay/threading

@@ -323,7 +323,7 @@ Fixing Threading
      
      When performing a state change on an element that returns ASYNC on one of
      the state changes, ASYNC is returned and you can only proceed to the next
-     state change change when this ASYNC state change completed. Use the
+     state change when this ASYNC state change completed. Use the
      gst_element_get_state function to know when the state change completed.
      An example of this behaviour is setting a videosink to PLAYING, it will
      return ASYNC in the state change from READY->PAUSED. You can only set

docs/random/wtay/threads_hilevel

@@ -168,5 +168,5 @@ TODO
 ----
 
 - review
-- figure all all state change scenarios occuring in code marked with (*)
+- figure all state change scenarios occuring in code marked with (*)
 

gst/gstbin.c

@@ -1859,7 +1859,7 @@ clear_queue (GQueue * queue)
 }
 
 /* set all degrees to 0. Elements marked as a sink are
- * added to the queue immediatly. Since we only look at the SINK flag of the
+ * added to the queue immediately. Since we only look at the SINK flag of the
  * element, it is possible that we add non-sinks to the queue. These will be
  * removed from the queue again when we can prove that it provides data for some
  * other element. */
@@ -2481,7 +2481,7 @@ gst_bin_change_state_func (GstElement * element, GstStateChange transition)
       break;
   }
 
-  /* this flag is used to make the async state changes return immediatly. We
+  /* this flag is used to make the async state changes return immediately. We
    * don't want them to interfere with this state change */
   GST_OBJECT_LOCK (bin);
   bin->polling = TRUE;
@@ -2633,7 +2633,7 @@ done:
     /* nothing found, remove all old ASYNC_DONE messages. This can happen when
      * all the elements commited their state while we were doing the state
      * change. We will still return ASYNC for consistency but we commit the
-     * state already so that a _get_state() will return immediatly. */
+     * state already so that a _get_state() will return immediately. */
     bin_remove_messages (bin, NULL, GST_MESSAGE_ASYNC_DONE);
 
     GST_DEBUG_OBJECT (bin, "async elements commited");
@@ -2741,7 +2741,7 @@ gst_bin_continue_func (BinContinueData * data)
   GST_OBJECT_LOCK (bin);
 
   /* if a new state change happened after this thread was scheduled, we return
-   * immediatly. */
+   * immediately. */
   if (data->cookie != GST_ELEMENT_CAST (bin)->state_cookie)
     goto interrupted;
 

gst/gstcaps.c

@@ -419,7 +419,7 @@ gst_caps_ref (GstCaps * caps)
  * gst_caps_unref:
  * @caps: (transfer full): the #GstCaps to unref
  *
- * Unref a #GstCaps and and free all its structures and the
+ * Unref a #GstCaps and free all its structures and the
  * structures' values when the refcount reaches 0.
  */
 void
@@ -707,7 +707,7 @@ gst_caps_remove_structure (GstCaps * caps, guint idx)
 
 /**
  * gst_caps_merge_structure:
- * @caps: the #GstCaps that will the the new structure
+ * @caps: the #GstCaps that will the new structure
  * @structure: (transfer full): the #GstStructure to merge
  *
  * Appends @structure to @caps if its not already expressed by @caps.  The
@@ -1497,8 +1497,8 @@ gst_caps_structure_subtract (GSList ** into, const GstStructure * minuend,
 
 /**
  * gst_caps_subtract:
- * @minuend: #GstCaps to substract from
+ * @minuend: #GstCaps to subtract from
- * @subtrahend: #GstCaps to substract
+ * @subtrahend: #GstCaps to subtract
  *
  * Subtracts the @subtrahend from the @minuend.
  * <note>This function does not work reliably if optional properties for caps

gst/gstchildproxy.c

@@ -229,7 +229,7 @@ gst_child_proxy_lookup (GstObject * object, const gchar * name,
  * @value: (out caller-allocates): a #GValue that should take the result.
  *
  * Gets a single property using the GstChildProxy mechanism.
- * You are responsible for for freeing it by calling g_value_unset()
+ * You are responsible for freeing it by calling g_value_unset()
  */
 void
 gst_child_proxy_get_property (GstObject * object, const gchar * name,

gst/gstelement.c

@@ -2128,7 +2128,7 @@ gst_element_get_state_func (GstElement * element,
   if (ret == GST_STATE_CHANGE_FAILURE)
     goto done;
 
-  /* we got no_preroll, report immediatly */
+  /* we got no_preroll, report immediately */
   if (ret == GST_STATE_CHANGE_NO_PREROLL)
     goto done;
 

gst/gstevent.c

@@ -50,7 +50,7 @@
  * Most of the event API is used inside plugins. Applications usually only 
  * construct and use seek events. 
  * To do that gst_event_new_seek() is used to create a seek event. It takes
- * the needed parameters to specity seeking time and mode.
+ * the needed parameters to specify seeking time and mode.
  * <example>
  * <title>performing a seek on a pipeline</title>
  *   <programlisting>
@@ -729,7 +729,7 @@ gst_event_parse_tag (GstEvent * event, GstTagList ** taglist)
  * Create a new buffersize event. The event is sent downstream and notifies
  * elements that they should provide a buffer of the specified dimensions.
  *
- * When the @async flag is set, a thread boundary is prefered.
+ * When the @async flag is set, a thread boundary is preferred.
  *
  * Returns: (transfer full): a new #GstEvent
  */

gst/gstevent.h

@@ -292,7 +292,7 @@ typedef enum {
  * no EOS will be emmited by the element that performed the seek, but a
  * #GST_MESSAGE_SEGMENT_DONE message will be posted on the bus by the element.
  * When this message is posted, it is possible to send a new seek event to
- * continue playback. With this seek method it is possible to perform seemless
+ * continue playback. With this seek method it is possible to perform seamless
  * looping or simple linear editing.
  *
  * When doing fast forward (rate > 1.0) or fast reverse (rate < -1.0) trickmode

gst/gstghostpad.c

@@ -615,7 +615,7 @@ gst_proxy_pad_unlink_default (GstPad * pad)
 
   /* don't do anything if this unlink resulted from retargeting the pad
    * controlled by the ghostpad. We only want to invalidate the target pad when
-   * the element suddently unlinked with our internal pad. */
+   * the element suddenly unlinked with our internal pad. */
   if (GST_PROXY_PAD_RETARGET (pad))
     return;
 
@@ -845,7 +845,7 @@ gst_ghost_pad_internal_activate_pull_default (GstPad * pad, gboolean active)
   if (GST_PAD_DIRECTION (pad) == GST_PAD_SRC) {
     /* we are activated in pull mode by our peer element, which is a sinkpad
      * that wants to operate in pull mode. This activation has to propagate
-     * upstream throught the pipeline. We call the internal activation function,
+     * upstream through the pipeline. We call the internal activation function,
      * which will trigger gst_ghost_pad_activate_pull_default, which propagates even
      * further upstream */
     GST_LOG_OBJECT (pad, "pad is src, activate internal");

gst/gstinterface.c

@@ -24,7 +24,7 @@
 /**
  * SECTION:gstimplementsinterface
  * @short_description: Core interface implemented by #GstElement instances that
- * allows runtime querying of interface availabillity
+ * allows runtime querying of interface availability
  * @see_also: #GstElement
  *
  * Provides interface functionality on per instance basis and not per class

gst/gstpad.c

@@ -2289,7 +2289,7 @@ done:
 }
 
 /* FIXME-0.11: what about making this the default and using
- * gst_caps_make_writable() explicitely where needed
+ * gst_caps_make_writable() explicitly where needed
  */
 /**
  * gst_pad_get_caps_reffed:
@@ -2348,7 +2348,7 @@ gst_pad_get_caps (GstPad * pad)
 }
 
 /* FIXME-0.11: what about making this the default and using
- * gst_caps_make_writable() explicitely where needed
+ * gst_caps_make_writable() explicitly where needed
  */
 /**
  * gst_pad_peer_get_caps_reffed:
@@ -3423,7 +3423,7 @@ gst_pad_get_internal_links_default (GstPad * pad)
 
     it = gst_pad_iterate_internal_links (pad);
     /* loop over the iterator and put all elements into a list, we also
-     * immediatly unref them, which is bad. */
+     * immediately unref them, which is bad. */
     do {
       ires = gst_iterator_foreach (it, (GFunc) add_unref_pad_to_list, &res);
       switch (ires) {
@@ -3616,7 +3616,7 @@ no_iter:
  * pads that are internally linked to @pad, only one will be sent an event.
  * Multi-sinkpad elements should implement custom event handlers.
  *
- * Returns: TRUE if the event was sent succesfully.
+ * Returns: TRUE if the event was sent successfully.
  */
 gboolean
 gst_pad_event_default (GstPad * pad, GstEvent * event)
@@ -3810,7 +3810,7 @@ no_peer:
  * @pad, only one will be sent the query.
  * Multi-sinkpad elements should implement custom query handlers.
  *
- * Returns: TRUE if the query was performed succesfully.
+ * Returns: TRUE if the query was performed successfully.
  */
 gboolean
 gst_pad_query_default (GstPad * pad, GstQuery * query)
@@ -4043,7 +4043,7 @@ handle_pad_block (GstPad * pad)
       pad->abidata.ABI.block_callback_called = TRUE;
       if (callback) {
         /* there is a callback installed, call it. We release the
-         * lock so that the callback can do something usefull with the
+         * lock so that the callback can do something useful with the
          * pad */
         user_data = pad->block_data;
         GST_OBJECT_UNLOCK (pad);
@@ -5015,7 +5015,7 @@ not_negotiated:
  *     returns #GST_FLOW_ERROR if %NULL.
  *
  * When @pad is flushing this function returns #GST_FLOW_WRONG_STATE
- * immediatly and @buffer is %NULL.
+ * immediately and @buffer is %NULL.
  *
  * Calls the getrange function of @pad, see #GstPadGetRangeFunction for a
  * description of a getrange function. If @pad has no getrange function
@@ -5376,7 +5376,7 @@ gst_pad_send_event (GstPad * pad, GstEvent * event)
           GST_EVENT_TYPE_NAME (event));
 
       /* make this a little faster, no point in grabbing the lock
-       * if the pad is allready flushing. */
+       * if the pad is already flushing. */
       if (G_UNLIKELY (GST_PAD_IS_FLUSHING (pad)))
         goto flushing;
 

gst/gstpad.h

@@ -171,9 +171,9 @@ typedef enum {
  * @ret: a #GstFlowReturn value
  *
  * Macro to test if the given #GstFlowReturn value indicates a
- * successfull result
+ * successful result
  * This macro is mainly used in elements to decide if the processing
- * of a buffer was successfull.
+ * of a buffer was successful.
  *
  * Since: 0.10.7
  *
@@ -510,7 +510,7 @@ typedef gboolean		(*GstPadAcceptCapsFunction)	(GstPad *pad, GstCaps *caps);
  * @pad: a #GstPad
  * @caps: the #GstCaps to fixate
  *
- * Given possibly unfixed caps @caps, let @pad use its default prefered
+ * Given possibly unfixed caps @caps, let @pad use its default preferred
  * format to make a fixed caps. @caps should be writable. By default this
  * function will pick the first value of any ranges or lists in the caps but
  * elements can override this function to perform other behaviour.

gst/gstparamspecs.h

@@ -1,4 +1,4 @@
-/* GStreamer - GParamSpecs for for some of our types
+/* GStreamer - GParamSpecs for some of our types
  * Copyright (C) 2007 Tim-Philipp Müller  <tim centricular net>
  *
  * This library is free software; you can redistribute it and/or

gst/gstparse.c

@@ -31,7 +31,7 @@
  *
  * Please note that these functions take several measures to create
  * somewhat dynamic pipelines. Due to that such pipelines are not always
- * reuseable (set the state to NULL and back to PLAYING).
+ * reusable (set the state to NULL and back to PLAYING).
  */
 
 #include "gst_private.h"

gst/gstpipeline.c

@@ -29,7 +29,7 @@
  * A #GstPipeline is a special #GstBin used as the toplevel container for
  * the filter graph. The #GstPipeline will manage the selection and
  * distribution of a global #GstClock as well as provide a #GstBus to the
- * application. It will also implement a default behavour for managing
+ * application. It will also implement a default behaviour for managing
  * seek events (see gst_element_seek()).
  *
  * gst_pipeline_new() is used to create a pipeline. when you are done with

gst/gstplugin.c

@@ -1764,7 +1764,7 @@ gst_plugin_ext_dep_equals (GstPluginDep * dep, const gchar ** env_vars,
 /**
  * gst_plugin_add_dependency:
  * @plugin: a #GstPlugin
- * @env_vars: NULL-terminated array of environent variables affecting the
+ * @env_vars: NULL-terminated array of environment variables affecting the
  *     feature set of the plugin (e.g. an environment variable containing
  *     paths where to look for additional modules/plugins of a library),
  *     or NULL. Environment variable names may be followed by a path component
@@ -1837,7 +1837,7 @@ gst_plugin_add_dependency (GstPlugin * plugin, const gchar ** env_vars,
 /**
  * gst_plugin_add_dependency_simple:
  * @plugin: the #GstPlugin
- * @env_vars: one or more environent variables (separated by ':', ';' or ','),
+ * @env_vars: one or more environment variables (separated by ':', ';' or ','),
  *      or NULL. Environment variable names may be followed by a path component
  *      which will be added to the content of the environment variable, e.g.
  *      "HOME/.mystuff/plugins:MYSTUFF_PLUGINS_PATH"

gst/gstpluginfeature.c

@@ -372,7 +372,7 @@ gst_plugin_feature_check_version (GstPluginFeature * feature,
         ret = FALSE;
       else if (micro > min_micro)
         ret = TRUE;
-      /* micro is 1 smaller but we have a nano version, this is the upcomming
+      /* micro is 1 smaller but we have a nano version, this is the upcoming
        * release of the requested version and we're ok then */
       else if (nscan == 4 && nano > 0 && (micro + 1 == min_micro))
         ret = TRUE;

gst/gstpluginfeature.h

@@ -52,7 +52,7 @@ typedef struct _GstPluginFeatureClass GstPluginFeatureClass;
 /**
  * GstRank:
  * @GST_RANK_NONE: will be chosen last or not at all
- * @GST_RANK_MARGINAL: unlikly to be chosen
+ * @GST_RANK_MARGINAL: unlikely to be chosen
  * @GST_RANK_SECONDARY: likely to be chosen
  * @GST_RANK_PRIMARY: will be chosen first
  *

gst/gstpoll.c

@@ -1232,7 +1232,7 @@ gst_poll_wait (GstPoll * set, GstClockTime timeout)
   if (G_UNLIKELY (old_waiting > 0 && !is_timer))
     goto already_waiting;
 
-  /* flushing, exit immediatly */
+  /* flushing, exit immediately */
   if (G_UNLIKELY (IS_FLUSHING (set)))
     goto flushing;
 

gst/gstregistry.c

@@ -103,7 +103,7 @@
  * different sets of plugins. For various reasons, at init time, the cache is
  * stored in the default registry, and plugins not relevant to the current
  * process are marked with the %GST_PLUGIN_FLAG_CACHED bit. These plugins are
- * removed at the end of intitialization.
+ * removed at the end of initialization.
  */
 
 #ifdef HAVE_CONFIG_H

gst/gststructure.c

@@ -2842,7 +2842,7 @@ wrong_type:
  * For refcounted (mini)objects you will acquire your own reference which
  * you must release with a suitable _unref() when no longer needed. For
  * strings and boxed types you will acquire a copy which you will need to
- * release with either g_free() or the suiteable function for the boxed type.
+ * release with either g_free() or the suitable function for the boxed type.
  *
  * Returns: FALSE if there was a problem reading any of the fields (e.g.
  *     because the field requested did not exist, or was of a type other
@@ -2887,7 +2887,7 @@ gst_structure_get (const GstStructure * structure, const char *first_fieldname,
  * For refcounted (mini)objects you will acquire your own reference which
  * you must release with a suitable _unref() when no longer needed. For
  * strings and boxed types you will acquire a copy which you will need to
- * release with either g_free() or the suiteable function for the boxed type.
+ * release with either g_free() or the suitable function for the boxed type.
  *
  * Returns: FALSE if there was a problem reading any of the fields (e.g.
  *     because the field requested did not exist, or was of a type other
@@ -3075,7 +3075,7 @@ gst_caps_structure_can_intersect_field (GQuark id, const GValue * val1,
  * @struct1: a #GstStructure
  * @struct2: a #GstStructure
  *
- * Tries interesecting @struct1 and @struct2 and reports whether the result
+ * Tries intersecting @struct1 and @struct2 and reports whether the result
  * would not be empty.
  *
  * Returns: %TRUE if intersection would not be empty

gst/gstutils.c

@@ -485,7 +485,7 @@ _gst_util_uint64_scale (guint64 val, guint64 num, guint64 denom,
   if (G_UNLIKELY (num == denom))
     return val;
 
-  /* on 64bits we always use a full 128bits multipy/division */
+  /* on 64bits we always use a full 128bits multiply/division */
 #if !defined (__x86_64__) && !defined (HAVE_UINT128_T)
   /* denom is low --> try to use 96 bit muldiv */
   if (G_LIKELY (denom <= G_MAXUINT32)) {
@@ -829,7 +829,7 @@ gst_print_element_args (GString * buf, gint indent, GstElement * element)
  * @element: (transfer none): a #GstElement to create pads for
  *
  * Creates a pad for each pad template that is always available.
- * This function is only useful during object intialization of
+ * This function is only useful during object initialization of
  * subclasses of #GstElement.
  */
 void
@@ -2045,7 +2045,7 @@ gst_element_link_pads_filtered (GstElement * src, const gchar * srcpadname,
  * Links @src to @dest. The link must be from source to
  * destination; the other direction will not be tried. The function looks for
  * existing pads that aren't linked yet. It will request new pads if necessary.
- * Such pads need to be released manualy when unlinking.
+ * Such pads need to be released manually when unlinking.
  * If multiple links are possible, only one is established.
  *
  * Make sure you have added your elements to a bin or pipeline with
@@ -2166,7 +2166,7 @@ gst_element_unlink_pads (GstElement * src, const gchar * srcpadname,
     goto free_src;
   }
 
-  /* we're satisified they can be unlinked, let's do it */
+  /* we're satisfied they can be unlinked, let's do it */
   gst_pad_unlink (srcpad, destpad);
 
   if (destrequest)
@@ -2956,7 +2956,7 @@ setcaps_fold_func (GstPad * pad, GValue * ret, SetCapsFoldData * data)
  * same element as @pad.  If gst_pad_set_caps() fails on any pad,
  * the proxy setcaps fails. May be used only during negotiation.
  *
- * Returns: TRUE if sucessful
+ * Returns: TRUE if successful
  */
 gboolean
 gst_pad_proxy_setcaps (GstPad * pad, GstCaps * caps)
@@ -3854,7 +3854,7 @@ gst_parse_bin_from_description_full (const gchar * bin_description,
  * @base_init: Location of the base initialization function (optional).
  * @base_finalize: Location of the base finalization function (optional).
  * @class_init: Location of the class initialization function for class types
- *   Location of the default vtable inititalization function for interface
+ *   Location of the default vtable initialization function for interface
  *   types. (optional)
  * @class_finalize: Location of the class finalization function for class types.
  *   Location of the default vtable finalization function for interface types.
@@ -3918,7 +3918,7 @@ gst_type_register_static_full (GType parent_type,
 /**
  * gst_util_get_timestamp:
  *
- * Get a timestamp as GstClockTime to be used for interval meassurements.
+ * Get a timestamp as GstClockTime to be used for interval measurements.
  * The timestamp should not be interpreted in any other way.
  *
  * Returns: the timestamp

gst/gstutils.h

@@ -505,7 +505,7 @@ GST_BOILERPLATE_FULL (type, type_as_function, parent_type,              \
 					  _GST_PUT (data, 0,  8,  0, num); \
 					} while (0)
 
-/* Float endianess conversion macros */
+/* Float endianness conversion macros */
 
 /* FIXME: Remove this once we depend on a GLib version with this */
 #ifndef GFLOAT_FROM_LE

gst/gstvalue.c

@@ -2295,7 +2295,7 @@ gst_string_unwrap (const gchar * s)
 
         read += 3;
       } else {
-        /* if we run into a \0 here, we definately won't get a quote later */
+        /* if we run into a \0 here, we definitely won't get a quote later */
         if (*read == 0)
           goto beach;
 
@@ -3542,7 +3542,7 @@ gst_value_union (GValue * dest, const GValue * value1, const GValue * value2)
  * gst_value_register_union_func:
  * @type1: a type to union
  * @type2: another type to union
- * @func: a function that implments creating a union between the two types
+ * @func: a function that implements creating a union between the two types
  *
  * Registers a union function that can create a union between #GValue items
  * of the type @type1 and @type2.
@@ -4374,7 +4374,7 @@ gst_value_compare_fraction (const GValue * value1, const GValue * value2)
  * @value: a GValue initialized to GST_TYPE_DATE
  * @date: the date to set the value to
  *
- * Sets the contents of @value to coorespond to @date.  The actual
+ * Sets the contents of @value to correspond to @date.  The actual
  * #GDate structure is copied before it is used.
  */
 void

libs/gst/base/gstbasesink.c

@@ -739,7 +739,7 @@ gst_base_sink_finalize (GObject * object)
  * @sync: the new sync value.
  *
  * Configures @sink to synchronize on the clock or not. When
- * @sync is FALSE, incomming samples will be played as fast as
+ * @sync is FALSE, incoming samples will be played as fast as
  * possible. If @sync is TRUE, the timestamps of the incomming
  * buffers will be used to schedule the exact render time of its
  * contents.
@@ -876,8 +876,8 @@ gst_base_sink_is_qos_enabled (GstBaseSink * sink)
  * @enabled: the new async value.
  *
  * Configures @sink to perform all state changes asynchronusly. When async is
- * disabled, the sink will immediatly go to PAUSED instead of waiting for a
+ * disabled, the sink will immediately go to PAUSED instead of waiting for a
- * preroll buffer. This feature is usefull if the sink does not synchronize
+ * preroll buffer. This feature is useful if the sink does not synchronize
  * against the clock or when it is dealing with sparse streams.
  *
  * Since: 0.10.15
@@ -2444,10 +2444,10 @@ flushing:
  * if needed and then block if we still are not PLAYING.
  *
  * We start waiting on the clock in PLAYING. If we got interrupted, we
- * immediatly try to re-preroll.
+ * immediately try to re-preroll.
  *
  * Some objects do not need synchronisation (most events) and so this function
- * immediatly returns GST_FLOW_OK.
+ * immediately returns GST_FLOW_OK.
  *
  * for objects that arrive later than max-lateness to be synchronized to the
  * clock have the @late boolean set to TRUE.
@@ -2567,7 +2567,7 @@ again:
       GST_TIME_FORMAT ", adjusted %" GST_TIME_FORMAT,
       GST_TIME_ARGS (rstart), GST_TIME_ARGS (stime));
 
-  /* This function will return immediatly if start == -1, no clock
+  /* This function will return immediately if start == -1, no clock
    * or sync is disabled with GST_CLOCK_BADTIME. */
   status = gst_base_sink_wait_clock (basesink, stime, &jitter);
 
@@ -2974,7 +2974,7 @@ again:
   step_end = FALSE;
 
   /* synchronize this object, non syncable objects return OK
-   * immediatly. */
+   * immediately. */
   ret =
       gst_base_sink_do_sync (basesink, pad, sync_obj, &late, &step_end,
       obj_type);
@@ -3264,7 +3264,7 @@ gst_base_sink_queue_object_unlocked (GstBaseSink * basesink, GstPad * pad,
       if (G_UNLIKELY (ret != GST_FLOW_OK))
         goto preroll_failed;
     }
-    /* need to recheck if we need preroll, commmit state during preroll
+    /* need to recheck if we need preroll, commit state during preroll
      * could have made us not need more preroll. */
     if (G_UNLIKELY (basesink->need_preroll)) {
       /* see if we can render now, if we can't add the object to the preroll
@@ -3910,7 +3910,7 @@ gst_base_sink_perform_seek (GstBaseSink * sink, GstPad * pad, GstEvent * event)
 
   /* If we configured the seeksegment above, don't overwrite it now. Otherwise
    * copy the current segment info into the temp segment that we can actually
-   * attempt the seek with. We only update the real segment if the seek suceeds. */
+   * attempt the seek with. We only update the real segment if the seek succeeds. */
   if (!seekseg_configured) {
     memcpy (&seeksegment, &sink->segment, sizeof (GstSegment));
 
@@ -3966,7 +3966,7 @@ gst_base_sink_perform_seek (GstBaseSink * sink, GstPad * pad, GstEvent * event)
     res = FALSE;
   }
 
-  /* if successfull seek, we update our real segment and push
+  /* if successful seek, we update our real segment and push
    * out the new segment. */
   if (res) {
     memcpy (&sink->segment, &seeksegment, sizeof (GstSegment));
@@ -4346,7 +4346,7 @@ gst_base_sink_negotiate_pull (GstBaseSink * basesink)
   GST_DEBUG_OBJECT (basesink, "allowed caps: %" GST_PTR_FORMAT, caps);
 
   caps = gst_caps_make_writable (caps);
-  /* get the first (prefered) format */
+  /* get the first (preferred) format */
   gst_caps_truncate (caps);
   /* try to fixate */
   gst_pad_fixate_caps (GST_BASE_SINK_PAD (basesink), caps);
@@ -5111,7 +5111,7 @@ gst_base_sink_change_state (GstElement * element, GstStateChange transition)
       break;
     case GST_STATE_CHANGE_PAUSED_TO_READY:
       GST_PAD_PREROLL_LOCK (basesink->sinkpad);
-      /* start by reseting our position state with the object lock so that the
+      /* start by resetting our position state with the object lock so that the
        * position query gets the right idea. We do this before we post the
        * messages so that the message handlers pick this up. */
       GST_OBJECT_LOCK (basesink);

libs/gst/base/gstbasesrc.c

@@ -1066,7 +1066,7 @@ gst_base_src_default_query (GstBaseSrc * src, GstQuery * query)
       GstClockTime min, max;
       gboolean live;
 
-      /* Subclasses should override and implement something usefull */
+      /* Subclasses should override and implement something useful */
       res = gst_base_src_query_latency (src, &live, &min, &max);
 
       GST_LOG_OBJECT (src, "report latency: live %d, min %" GST_TIME_FORMAT
@@ -1398,7 +1398,7 @@ gst_base_src_perform_seek (GstBaseSrc * src, GstEvent * event, gboolean unlock)
 
   /* If we configured the seeksegment above, don't overwrite it now. Otherwise
    * copy the current segment info into the temp segment that we can actually
-   * attempt the seek with. We only update the real segment if the seek suceeds. */
+   * attempt the seek with. We only update the real segment if the seek succeeds. */
   if (!seekseg_configured) {
     memcpy (&seeksegment, &src->segment, sizeof (GstSegment));
 
@@ -1587,7 +1587,7 @@ gst_base_src_send_event (GstElement * element, GstEvent * event)
        *    first and do EOS instead of entering it.
        *  - If we are in the _create function or we did not manage to set the
        *    flag fast enough and we are about to enter the _create function,
-       *    we unlock it so that we exit with WRONG_STATE immediatly. We then
+       *    we unlock it so that we exit with WRONG_STATE immediately. We then
        *    check the EOS flag and do the EOS logic.
        */
       g_atomic_int_set (&src->priv->pending_eos, TRUE);
@@ -3152,7 +3152,7 @@ gst_base_src_change_state (GstElement * element, GstStateChange transition)
        * already did this */
 
       /* FIXME, deprecate this behaviour, it is very dangerous.
-       * the prefered way of sending EOS downstream is by sending
+       * the preferred way of sending EOS downstream is by sending
        * the EOS event to the element */
       if (!basesrc->priv->last_sent_eos) {
         GST_DEBUG_OBJECT (basesrc, "Sending EOS event");

libs/gst/base/gstbasetransform.c

@@ -1546,7 +1546,7 @@ gst_base_transform_prepare_output_buffer (GstBaseTransform * trans,
        * it needs to revisit the decision about whether to proxy or not: */
       gst_caps_replace (&priv->sink_alloc, NULL);
       /* if we got a buffer of the wrong size, discard it now and make sure we
-       * allocate a propertly sized buffer later. */
+       * allocate a properly sized buffer later. */
       if (newsize != expsize) {
         if (in_buf != *out_buf)
           gst_buffer_unref (*out_buf);

libs/gst/base/gsttypefindhelper.c

@@ -150,7 +150,7 @@ helper_find_peek (gpointer data, gint64 offset, guint size)
   buf_size = GST_BUFFER_SIZE (buffer);
 
   if ((buf_offset != -1 && buf_offset != offset) || buf_size < size) {
-    GST_DEBUG ("droping short buffer: %" G_GUINT64_FORMAT "-%" G_GUINT64_FORMAT
+    GST_DEBUG ("dropping short buffer: %" G_GUINT64_FORMAT "-%" G_GUINT64_FORMAT
         " instead of %" G_GUINT64_FORMAT "-%" G_GUINT64_FORMAT,
         buf_offset, buf_offset + buf_size - 1, offset, offset + size - 1);
     gst_buffer_unref (buffer);
@@ -191,7 +191,7 @@ helper_find_suggest (gpointer data, GstTypeFindProbability probability,
   GstTypeFindHelper *helper = (GstTypeFindHelper *) data;
 
   GST_LOG_OBJECT (helper->obj,
-      "'%s' called called suggest (%u, %" GST_PTR_FORMAT ")",
+      "'%s' called suggest (%u, %" GST_PTR_FORMAT ")",
       GST_PLUGIN_FEATURE_NAME (helper->factory), probability, caps);
 
   if (probability > helper->best_probability) {
@@ -208,7 +208,7 @@ helper_find_get_length (gpointer data)
 {
   GstTypeFindHelper *helper = (GstTypeFindHelper *) data;
 
-  GST_LOG_OBJECT (helper->obj, "'%s' called called get_length, returning %"
+  GST_LOG_OBJECT (helper->obj, "'%s' called get_length, returning %"
       G_GUINT64_FORMAT, GST_PLUGIN_FEATURE_NAME (helper->factory),
       helper->size);
 
@@ -465,7 +465,7 @@ buf_helper_find_suggest (gpointer data, GstTypeFindProbability probability,
   GstTypeFindBufHelper *helper = (GstTypeFindBufHelper *) data;
 
   GST_LOG_OBJECT (helper->obj,
-      "'%s' called called suggest (%u, %" GST_PTR_FORMAT ")",
+      "'%s' called suggest (%u, %" GST_PTR_FORMAT ")",
       GST_PLUGIN_FEATURE_NAME (helper->factory), probability, caps);
 
   /* Note: not >= as we call typefinders in order of rank, highest first */