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For each lib we build export its own API in headers when we're building it, otherwise import the API from the headers. This fixes linker warnings on Windows when building with MSVC. The problem was that we had defined all GST_*_API decorators unconditionally to GST_EXPORT. This was intentional and only supposed to be temporary, but caused linker warnings because we tell the linker that we want to export all symbols even those from externall DLLs, and when the linker notices that they were in external DLLS and not present locally it warns. What we need to do when building each library is: export the library's own symbols and import all other symbols. To this end we define e.g. BUILDING_GST_FOO and then we define the GST_FOO_API decorator either to export or to import symbols depending on whether BUILDING_GST_FOO is set or not. That way external users of each library API automatically get the import. While we're at it, add new GST_API_EXPORT in config.h and use that for GST_*_API decorators instead of GST_EXPORT. The right export define depends on the toolchain and whether we're using -fvisibility=hidden or not, so it's better to set it to the right thing directly than hard-coding a compiler whitelist in the public header. We put the export define into config.h instead of passing it via the command line to the compiler because it might contain spaces and brackets and in the autotools scenario we'd have to pass that through multiple layers of plumbing and Makefile/shell escaping and we're just not going to be *that* lucky. The export define is only used if we're compiling our lib, not by external users of the lib headers, so it's not a problem to put it into config.h Also, this means all .c files of libs need to include config.h to get the export marker defined, so fix up a few that didn't include config.h. This commit depends on a common submodule commit that makes gst-glib-gen.mak add an #include "config.h" to generated enum/marshal .c files for the autotools build. https://bugzilla.gnome.org/show_bug.cgi?id=797185 |
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| .. | ||
| gstrtcpbuffer.c | ||
| gstrtcpbuffer.h | ||
| gstrtpbaseaudiopayload.c | ||
| gstrtpbaseaudiopayload.h | ||
| gstrtpbasedepayload.c | ||
| gstrtpbasedepayload.h | ||
| gstrtpbasepayload.c | ||
| gstrtpbasepayload.h | ||
| gstrtpbuffer.c | ||
| gstrtpbuffer.h | ||
| gstrtpdefs.h | ||
| gstrtphdrext.c | ||
| gstrtphdrext.h | ||
| gstrtppayloads.c | ||
| gstrtppayloads.h | ||
| Makefile.am | ||
| meson.build | ||
| README | ||
| rtp-prelude.h | ||
| 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.