gstreamer/gst-libs/gst/rtp
Wim Taymans 86ab51207b gst-libs/gst/rtp/gstbasertpdepayload.c: Validate the RTP packet before further processing it. It's just too dangerous...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain):
Validate the RTP packet before further processing it. It's just too
dangerous to accept random packets and people are not forced to use a
jitterbuffer or session manager to filter out the bad packets.
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_set_extension_data),
(gst_rtp_buffer_get_payload_subbuffer):
Small cleanups.
When setting extension data in a buffer that is too small, we fail and
we should not set the extension bit.
Change GST_WARNINGS into g_warning because they really are
programming errors.
* tests/check/libs/rtp.c: (GST_START_TEST):
Catch the g_warnings now in the unit tests and that fact that failing to
set extension data left the extension bit untouched.
2008-05-14 20:28:02 +00:00
..
gstbasertpaudiopayload.c gst-libs/gst/rtp/gstbasertpaudiopayload.*: API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it po... 2008-03-03 16:11:50 +00:00
gstbasertpaudiopayload.h gst-libs/gst/rtp/gstbasertpaudiopayload.*: API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it po... 2008-03-03 16:11:50 +00:00
gstbasertpdepayload.c gst-libs/gst/rtp/gstbasertpdepayload.c: Validate the RTP packet before further processing it. It's just too dangerous... 2008-05-14 20:28:02 +00:00
gstbasertpdepayload.h gst-libs/gst/rtp/gstbasertpdepayload.*: Catch packet-lost events from the jitterbuffer and convert them into a vmetho... 2008-05-02 12:11:07 +00:00
gstbasertppayload.c gst-libs/gst/rtp/gstbasertppayload.c: Rename the setcaps/getcaps function internally to make it clear that they are c... 2008-05-02 12:13:08 +00:00
gstbasertppayload.h gst-libs/gst/rtp/gstbasertppayload.*: Add getcaps vfunc to basertppayload. See #465146. 2007-08-16 16:06:21 +00:00
gstrtcpbuffer.c Correct all relevant warnings found by the sparse semantic code analyzer. This include marking several symbols static... 2008-03-03 06:04:31 +00:00
gstrtcpbuffer.h Fix parsing of RB blocks. 2007-09-03 19:31:11 +00:00
gstrtpbuffer.c gst-libs/gst/rtp/gstbasertpdepayload.c: Validate the RTP packet before further processing it. It's just too dangerous... 2008-05-14 20:28:02 +00:00
gstrtpbuffer.h Add gst_rtp_buffer_set_extension_data() 2008-02-01 11:09:16 +00:00
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 gst-libs/gst/rtp/: Added new file and header to deal with payload info. 2007-10-01 13:22:14 +00:00
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.