mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-26 18:20:44 +00:00
b7facbaf22
When generating segment, we can't assume the first buffer is actually the first expected one. If it's not, we need to adjust the segment to start a bit before. Additionally, we if don't know when the stream is suppose to have started (no clock-base in caps), it means we need to keep everything in running time and only rely on jitterbuffer to synchronize. https://bugzilla.gnome.org/show_bug.cgi?id=635701 |
||
|---|---|---|
| .. | ||
| gstrtcpbuffer.c | ||
| gstrtcpbuffer.h | ||
| gstrtpbaseaudiopayload.c | ||
| gstrtpbaseaudiopayload.h | ||
| gstrtpbasedepayload.c | ||
| gstrtpbasedepayload.h | ||
| gstrtpbasepayload.c | ||
| gstrtpbasepayload.h | ||
| gstrtpbuffer.c | ||
| gstrtpbuffer.h | ||
| gstrtphdrext.c | ||
| gstrtphdrext.h | ||
| gstrtppayloads.c | ||
| gstrtppayloads.h | ||
| Makefile.am | ||
| README | ||
| rtp.h | ||
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. 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.