mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-08 13:18:52 +00:00
8675bc89e4
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.
90 lines
4.4 KiB
Plaintext
90 lines
4.4 KiB
Plaintext
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.
|