gstreamer/gst-libs/gst/rtp/README
Philippe Kalaf 8675bc89e4 gst-libs/gst/rtp/README: Some new documentation
Original commit message from CVS:
2006-05-18 Philippe Kalaf <philippe.kalaf@collabora.co.uk>

* gst-libs/gst/rtp/README:
Some new documentation
* gst-libs/gst/rtp/gstrtpbuffer.h:
Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
New RTP audio base payloader class. Supports frame or sample based codecs.
Not enabled in Makefile.am until approved.
2006-05-18 23:00:02 +00:00

89 lines
4.4 KiB
Text

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 class derives from GstBaseRTPPayload.
It can be used for payloading audio codecs. It will only work with constant
bitrate codecs. It supports both frame based and sample based codecs. It takes
care of packing up the audio data into RTP packets and filling up the headers
accordingly. The payloading is done based on the maximum MTU (mtu) and the
maximum time per packet (max-ptime). The general idea is to divide large data
buffers into smaller RTP packets. The RTP packet size is the minimum of either
the MTU, max-ptime (if set) or available data. Any residual data is always
sent in a last RTP packet (no minimum RTP packet size). The idea is that since
this is a real time protocol, data should never be delayed. In the case of
frame based codecs, the resulting RTP packets always contain full frames.
To use this base class, your child element needs to call either
gst_basertpaudiopayload_set_frame_based() or
gst_basertpaudiopayload_set_sample_based(). This is usually done in the
element's _init() function. Then, the child element must call either
gst_basertpaudiopayload_set_frame_options() or
gst_basertpaudiopayload_set_sample_options(). Since GstBaseRTPAudioPayload
derives from GstBaseRTPPayload, the child element must set any variables or
call/override any functions required by that base class. The child element
does not need to override any other functions specific to
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.