gstreamer/gst-libs/gst/rtp
Alessandro Decina 030f639a8e android: make it ready for androgenizer
Remove the android/ top dir
Fixe the Makefile.am to be androgenized

To build gstreamer for android we are now using androgenizer which generates the
needed Android.mk files.
Androgenizer can be found here:
http://git.collabora.co.uk/?p=user/derek/androgenizer.git
2011-04-11 07:23:21 +02:00
..
gstbasertpaudiopayload.c baseaudiopay: fix timestamps on buffer lists 2010-12-30 18:09:58 +01:00
gstbasertpaudiopayload.h audiortppayload: refactor some more 2009-09-03 17:58:59 +02:00
gstbasertpdepayload.c rtp: Unref events if the parent element disappeared or has no event handler implemented 2011-04-08 15:10:02 +02:00
gstbasertpdepayload.h depay: update some docs 2010-12-21 15:11:10 +01:00
gstbasertppayload.c rtp: Unref events if the parent element disappeared or has no event handler implemented 2011-04-08 15:10:02 +02:00
gstbasertppayload.h audiopayload: add property to control packet duration 2010-04-09 16:17:31 +02:00
gstrtcpbuffer.c rtcpbuffer: Round to next 32bit word, not current 32bit word at end of SDES chunk 2011-04-05 15:27:03 +02:00
gstrtcpbuffer.h rtcpbuffer: Add function to manipulation the data in RTCP feedback packets 2010-10-05 16:19:14 +02:00
gstrtpbuffer.c rtp: Remove unused variables 2011-04-07 10:16:39 +02:00
gstrtpbuffer.h rtpbuffer: Add function to transform a GstBuffer into a GstBufferList 2010-10-05 16:19:14 +02:00
gstrtppayloads.c gst-libs/gst/rtp/gstrtppayloads.c: Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp(). 2008-04-19 16:33:24 +00:00
gstrtppayloads.h gst-libs/gst/rtp/: Added new file and header to deal with payload info. 2007-10-01 13:22:14 +00:00
Makefile.am android: make it ready for androgenizer 2011-04-11 07:23:21 +02:00
README gst-libs/gst/rtp/: Moved some documentation into .c file 2006-09-29 23:50:53 +00:00

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.