Based upon valgrind finding:
Conditional jump or move depends on uninitialised value(s)
at 0x4AFF589: read_rtp_header_extensions (gstrtpbasedepayload.c:1197)
by 0x4AFF9E5: gst_rtp_base_depayload_set_headers
(gstrtpbasedepayload.c:1298)
by 0x4AFFEE0: gst_rtp_base_depayload_do_push
(gstrtpbasedepayload.c:1413)
by 0x4AFFF53: gst_rtp_base_depayload_push
(gstrtpbasedepayload.c:1448)
by 0x4AFDEBA: gst_rtp_base_depayload_handle_buffer
(gstrtpbasedepayload.c:801)
by 0x4AFE41E: gst_rtp_base_depayload_chain_list
(gstrtpbasedepayload.c:899)
by 0x48F262C: gst_pad_chain_data_unchecked (gstpad.c:4414)
by 0x48F3333: gst_pad_push_data (gstpad.c:4655)
by 0x48F3DF8: gst_pad_push_list (gstpad.c:4814)
by 0x4AFAD87: gst_rtp_base_payload_push_list
(gstrtpbasepayload.c:1978)
by 0x72B3154: gst_rtp_vp8_pay_handle_buffer (gstrtpvp8pay.c:672)
by 0x4AF7031: gst_rtp_base_payload_chain (gstrtpbasepayload.c:868)
Uninitialised value was created by a heap allocation
at 0x483C77F: malloc (in
/usr/lib/x86_64-linux-gnu/valgrind/vgpreload_memcheck-amd64-linux.so)
by 0x4B8BA78: g_malloc (gmem.c:106)
by 0x4BA3A9D: g_slice_alloc (gslice.c:1069)
by 0x488D777: _sysmem_new_block (gstallocator.c:413)
by 0x488DB28: default_alloc (gstallocator.c:512)
by 0x488D3E8: gst_allocator_alloc (gstallocator.c:310)
by 0x4AE97E3: gst_rtp_buffer_set_extension_data (gstrtpbuffer.c:856)
by 0x4AF9EC6: set_headers (gstrtpbasepayload.c:1757)
by 0x489FE4D: gst_buffer_list_foreach (gstbufferlist.c:287)
by 0x4AFA87A: gst_rtp_base_payload_prepare_push
(gstrtpbasepayload.c:1915)
by 0x4AFAD06: gst_rtp_base_payload_push_list
(gstrtpbasepayload.c:1970)
by 0x72B3154: gst_rtp_vp8_pay_handle_buffer (gstrtpvp8pay.c:672)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1075>
We validate the header extensions length of an RTP buffer by comparing
it against the block size. Since we multiply the length in words by 4 to
get the length in bytes, a suitably large length could cause a wrapround
of the uint16, giving a lower length which erroneously passes the check
and allows the buffer to be mapped.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/864>
The wordlen ("length") MUST represent the total "number of 32-bit words
in the extension, excluding the four-octet extension header" (rfc3550).
There are cases where already existent padding is reused for adding
the new extension. So the new wordlen should be updated if the new
added extension makes it to increase.
If timestamp goes forwards more than allowed, we consider that the
timestamp belongs to the previous counting, so the extended timestamp
is unwrapped.
https://bugzilla.gnome.org/show_bug.cgi?id=783443
When gst_rtp_buffer_add_extension_onebyte_header() is used over a
GstRtpBuffer that only contains a memory for the whole packet,
ensure_buffers function crashes at the next point:
mem = gst_memory_copy (rtp->map[i].memory, offset, rtp->size[i]);
when i==2 because the payload is not mapped.
In addition the offset is calculated subtracting in the wrong direction.
https://bugzilla.gnome.org/show_bug.cgi?id=774959
gst_rtp_buffer_add_extension_onebyte_header() and
gst_rtp_buffer_add_extension_twobytes_header() can have a const argument for
the actual extension data.
Encrypted RTP buffers may contain encrypted padding, hence it's
necessary to have an option to relax the validation in order to
successfully map the buffer.
When the flag GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING is set
gst_rtp_buffer_map() will map the buffer like if padding is not
present.
https://bugzilla.gnome.org/show_bug.cgi?id=752705
Micro-optimisation: if the buffer consist of just one memory, we
know we have already mapped that memory to read the headers, so
no need to map it another time to get to the payload data, we
can just set up the payload data details right there and then
and avoid another map call in gst_rtp_buffer_get_payload().
Adds up when receiving RTP-payloaded raw video which can easily
be thousands of packets per frame.
They are very confusing for people, and more often than not
also just not very accurate. Seeing 'last reviewed: 2005' in
your docs is not very confidence-inspiring. Let's just remove
those comments.
The payload type can't be between 72 and 76 because with the marker bit set,
this could be mistaken for an RTCP packet then. We do a relaxed check and
only refuse 72-76 when the marker bit is set. The effect is that when
we try to map an RTCP packet as an RTP packet, we will certainly fail.
The function gst_rtp_buffer_get_payload can not be used in Python
because it lacks necessary length parameter. This patch adds a new
function, gst_rtp_buffer_get_payload_bytes, to use from Python
bindings. The new function has the advisory "Rename to:" annotation
so it can replace the gst_rtp_buffer_get_payload whan creating
bindings.
The function gst_rtp_buffer_get_extension_bytes is also added. It wraps
gst_rtp_buffer_get_extension_data which doesn't work in Python due to
incomplete annotation and because it returns the length as number of
32-bit words.
https://bugzilla.gnome.org/show_bug.cgi?id=698562
Allocate header, payload and padding in separate memory blocks in
gst_rtp_buffer_allocate().
don't use part of the payload data as storage for the extension data but store
it in a separate memory block that can be enlarged when needed.
Rework the one and two-byte header extension to make it reserve space for the
extra extension first.
Fix RTP unit test. Don't map the complete buffer or make assumptions on the
memory layout of the underlaying implementation. We can now always add extension
data because we have a separate memory block for it.
Add support RTP buffers with multiple memory blocks. We allow one block for the
header, one for the extension data, N for data and one memory block for the
padding.
Remove the validate function, we validate now when we map because we need to
parse things in order to map multiple memory blocks.
