gstreamer/gst-libs/gst/rtp
Wim Taymans 24347217a5 rtp: fix de/payloaders
gst_basertppayload -> gst_base_rtp_payload
Add pts/dts support in the depayloader
Remove old timestamp code
Add a default getcaps function so subclasses can chain up to it instead of
relying on the return value of the getcaps function.
2011-11-10 17:18:00 +01:00
..
gst-plugins-base-rtcp-feedback.patch Merge branch 'master' into 0.11 2011-09-06 15:24:32 +02:00
gstbasertpaudiopayload.c rtp: fix de/payloaders 2011-11-10 17:18:00 +01:00
gstbasertpaudiopayload.h docs: rtp library docs update 2011-05-23 23:56:09 +03:00
gstbasertpdepayload.c rtp: fix de/payloaders 2011-11-10 17:18:00 +01:00
gstbasertpdepayload.h rtp: fix de/payloaders 2011-11-10 17:18:00 +01:00
gstbasertppayload.c rtp: fix de/payloaders 2011-11-10 17:18:00 +01:00
gstbasertppayload.c.orig Merge branch 'master' into 0.11 2011-09-06 15:24:32 +02:00
gstbasertppayload.c.rej Merge branch 'master' into 0.11 2011-09-06 15:24:32 +02:00
gstbasertppayload.h rtp: fix de/payloaders 2011-11-10 17:18:00 +01:00
gstrtcpbuffer.c rtcpbuffer: provide a WRITE map with maximum available size 2011-07-09 18:23:18 +02:00
gstrtcpbuffer.h libs: port to new data API 2011-03-27 13:55:15 +02:00
gstrtpbuffer.c rtpbuffer: Fix compilation issues with gcc 4.6.1 2011-11-04 10:36:15 +01:00
gstrtpbuffer.c.new Merge branch 'master' into 0.11 2011-09-06 15:24:32 +02:00
gstrtpbuffer.h tests: fix RTP and RTCP unit tests 2011-03-28 18:42:09 +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 Add -DGST_USE_UNSTABLE_API to the compiler flags to avoid warnings 2011-07-05 10:07:08 +01:00
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.