--- a/gst_plugins_base/gst-libs/gst/rtp/README	Tue Aug 31 15:30:33 2010 +0300
+++ /dev/null	Thu Jan 01 00:00:00 1970 +0000
@@ -1,66 +0,0 @@
-The RTP libraries
----------------------
-
-  RTP Buffers
-  -----------
-  The real time protocol as described in RFC 3550 requires the use of special
-  packets containing an additional RTP header of at least 12 bytes. GStreamer
-  provides some helper functions for creating and parsing these RTP headers.
-  The result is a normal #GstBuffer with an additional RTP header.
- 
-  RTP buffers are usually created with gst_rtp_buffer_new_allocate() or
-  gst_rtp_buffer_new_allocate_len(). These functions create buffers with a
-  preallocated space of memory. It will also ensure that enough memory
-  is allocated for the RTP header. The first function is used when the payload
-  size is known. gst_rtp_buffer_new_allocate_len() should be used when the size
-  of the whole RTP buffer (RTP header + payload) is known.
- 
-  When receiving RTP buffers from a network, gst_rtp_buffer_new_take_data()
-  should be used when the user would like to parse that RTP packet. (TODO Ask
-  Wim what the real purpose of this function is as it seems to simply create a
-  duplicate GstBuffer with the same data as the previous one). The
-  function will create a new RTP buffer with the given data as the whole RTP
-  packet. Alternatively, gst_rtp_buffer_new_copy_data() can be used if the user
-  wishes to make a copy of the data before using it in the new RTP buffer. An
-  important function is gst_rtp_buffer_validate() that is used to verify that
-  the buffer a well formed RTP buffer.
- 
-  It is now possible to use all the gst_rtp_buffer_get_*() or
-  gst_rtp_buffer_set_*() functions to read or write the different parts of the
-  RTP header such as the payload type, the sequence number or the RTP
-  timestamp. The use can also retreive a pointer to the actual RTP payload data
-  using the gst_rtp_buffer_get_payload() function.
-
-  RTP Base Payloader Class (GstBaseRTPPayload)
-  --------------------------------------------
-
-  All RTP payloader elements (audio or video) should derive from this class.
-
-  RTP Base Audio Payloader Class (GstBaseRTPAudioPayload)
-  -------------------------------------------------------
-
-  This base class can be tested through it's children classes. Here is an
-  example using the iLBC payloader (frame based).
-
-  For 20ms mode :
-
-  GST_DEBUG="basertpaudiopayload:5" gst-launch-0.10 fakesrc sizetype=2
-  sizemax=114 datarate=1900 ! audio/x-iLBC, mode=20 !  rtpilbcpay
-  max-ptime="40000000" ! fakesink
-
-  For 30ms mode :
-
-  GST_DEBUG="basertpaudiopayload:5" gst-launch-0.10 fakesrc sizetype=2
-  sizemax=150 datarate=1662 ! audio/x-iLBC, mode=30 !  rtpilbcpay
-  max-ptime="60000000" ! fakesink
-
-  Here is an example using the uLaw payloader (sample based).
-
-  GST_DEBUG="basertpaudiopayload:5" gst-launch-0.10 fakesrc sizetype=2
-  sizemax=150 datarate=8000 ! audio/x-mulaw ! rtppcmupay max-ptime="6000000" !
-  fakesink
-
-  RTP Base Depayloader Class (GstBaseRTPDepayload)
-  ------------------------------------------------
-
-  All RTP depayloader elements (audio or video) should derive from this class.