Commit graph

890 commits

Author SHA1 Message Date
Luis de Bethencourt
823194284c rtph264depay: remove unused value
CID #1226474
2015-03-09 16:22:33 +00:00
Luis de Bethencourt
5cd293fe76 rtph263pay: fix leak
CID 1212156
2015-03-09 16:17:45 +00:00
Luis de Bethencourt
e87113781a rtph263pay: remove uneeded variable
We just need to save the ebit information in case there is an error decoding.
2015-03-09 16:17:45 +00:00
Sebastian Rasmussen
d331d931db rtpvorbispay: fix payloader description and author e-mail
https://bugzilla.gnome.org/show_bug.cgi?id=745226
2015-02-26 15:57:08 +00:00
Vincent Penquerc'h
dc73d153cb rtpvp8pay: default encoding name to VP8
https://bugzilla.gnome.org/show_bug.cgi?id=737810
2015-02-19 14:29:02 +00:00
Vincent Penquerc'h
b88ea286d2 rtpvp8pay: make caps writable before truncating them
https://bugzilla.gnome.org/show_bug.cgi?id=737810
2015-02-19 14:06:51 +00:00
Vincent Penquerc'h
b866c989f5 rtpvp8pay: negotiate encoding name
Chrome uses a different one than gstreamer.

https://bugzilla.gnome.org/show_bug.cgi?id=737810
2015-02-19 13:52:29 +00:00
Wim Taymans
009a62fddb rtph263depay: fix compilation with gcc 5.0 2015-02-10 18:54:24 +01:00
Thiago Santos
a6d73797d0 rtph264depay: prevent trying to get 0 bytes from adapter
This causes an assertion and would lead to getting a NULL instead
of a buffer. Without proper checking this would easily lead to
a segfault

https://bugzilla.gnome.org/show_bug.cgi?id=737199
2015-02-04 21:37:50 -03:00
Patrick Radizi
0a359cdbdc rtph264pay: fix potential crash when shutting down
A race condition in the state change function may cause buffers
to be unreffed while they are still used by the streaming thread
in gst_rtp_h264_pay_send_sps_pps() resulting in a crash. Chain
up to the parent class first in the state change function to
make sure streaming has stopped and only then free those buffers.

https://bugzilla.gnome.org/show_bug.cgi?id=741381
2014-12-11 14:00:19 +00:00
Patrick Radizi
fef1a8d88a rtph264pay: Fixes buffer leak when using SPS/PPS
Fixes a buffer leak that would occurr if the pipeline was shutdown
while a SPS/PPS header was being created.

https://bugzilla.gnome.org/show_bug.cgi?id=741271
2014-12-09 09:47:23 +01:00
Olivier Crête
e3b0fb2a5d rtpmpadepay: Relax caps to allow any clock-rate
Some Wowza setups seem to send an invalid non-90000 clock-rate.
2014-12-02 15:33:25 -05:00
Wim Taymans
3d7b0f30d7 rtpgstpay: put 0-byte at the end of events
Put a 0-byte at the end of the event string. Does not break ABI because
old depayloaders will skip the 0 byte (which is included in the length).
Expect a 0-byte at the end of the event string or a ; for old
payloaders.

See https://bugzilla.gnome.org/show_bug.cgi?id=737591
2014-11-20 13:14:14 +01:00
Wim Taymans
9d2902d978 rtpgstdepay: avoid buffer overread.
Check that a caps event string is 0 terminated and the event string is
terminated with a ; to avoid buffer overreads.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737591
2014-11-20 12:44:26 +01:00
Nicolas Dufresne
0f4f948f5f rtpvp8: Use VP8 encoding name
Both Firefox and Chrome uses VP8 as the encoding in their SDP.
Adding this now defacto standard name removes the need for special
case in SDP parsing code.

https://bugzilla.gnome.org/show_bug.cgi?id=737810
2014-11-01 11:26:26 -04:00
Tim-Philipp Müller
92c1d289b8 rtpmp2tpay: fix up template caps so we can output the default pt 33
Add fixed payload type for mp2t to template caps as well, so
our output caps match the advertised default pt. Fixes a
regression from 1.2.

There's still something wrong with caps negotiation though,
rtpmp2tpay payload=96 ! fakesink will not output caps with
payload=96.
2014-11-01 12:40:07 +00:00
Tim-Philipp Müller
f3fec86bc9 Revert "rtp: add h265 RTP payloader + depayloader"
This reverts commit d06ba9051f.

This breaks the build, as it depends on parser API in -bad.
2014-10-15 17:48:46 +01:00
Jurgen Slowack
d06ba9051f rtp: add h265 RTP payloader + depayloader 2014-10-15 17:34:50 +02:00
Sebastian Dröge
4bc10e755a rtpvrawdepay: Declare some more required caps fields in the sink template caps
Now only missing are width and height, which are expressed as strings
for RTP... so we can't put them into the template caps.
2014-09-16 22:47:13 +03:00
Sebastian Rasmussen
276269d956 rtph263ppay: Unref pad template caps after use
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734435
2014-08-08 16:02:24 -03:00
Srimanta Panda
421b00cd17 rtph264pay: append packetization mode parameter to SDP
Append packetization-mode parameter to SDP description.
Packetization mode signals the properties of an RTP payload type.

https://bugzilla.gnome.org/show_bug.cgi?id=733556
2014-08-08 13:41:36 +01:00
Mark Nauwelaerts
d5d28055c1 rtph264pay: unbreak au aligned byte-stream payloading 2014-08-03 14:42:45 +02:00
Srimanta Panda
dd9f716892 rtph264pay: append profile-level-id to SDP
Append profile-level-id to SDP if available.

https://bugzilla.gnome.org/show_bug.cgi?id=733539
2014-08-01 16:01:07 +01:00
Wim Taymans
8a78fa1ff5 vp8depay: fix header size checking
Use a different variable name to make it clear that we are calculating
the header size.
Correctly check that we have enough bytes to read the header bits. We
were checking if there were 5 bytes available in the header while we
only needed 3, causing the packet to be discarded as too small.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723595
2014-06-19 15:29:46 +02:00
Guillaume Desmottes
f00c2b7155 rtph264pay: propagate the GST_BUFFER_FLAG_DISCONT flag
Similarly to what we did with the DELTA_UNIT flag, this patch
propagates the DISCONT flag to the first RTP packet being used to transfer a
DISCONT buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=730563
2014-06-19 12:22:49 +02:00
Guillaume Desmottes
4be99ec7d5 rtph264pay: propagate the GST_BUFFER_FLAG_DELTA_UNIT flag
Downstream elements may be interested knowing if a RTP packet is the start
of a key frame (to implement a RTP extension as defined in the
ONVIF Streaming Spec for example).

We do this by checking the GST_BUFFER_FLAG_DELTA_UNIT flag we receive from
upstream and propagate it to the *first* RTP packet outputted to transfer this
buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=730563
2014-06-19 12:22:38 +02:00
Guillaume Desmottes
42ff642372 gstrtpmp4gpay: propagate the GST_BUFFER_FLAG_DISCONT flag
Propagate the DISCONT flag to the first RTP packet being used to transfer
a DISCONT buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=730563
2014-06-18 16:25:07 +02:00
Guillaume Desmottes
9a7479fb0d rtpjpegpay: propagate the GST_BUFFER_FLAG_DISCONT flag
Propagate the DISCONT flag to the first RTP packet being used to transfer
a DISCONT buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=730563
2014-06-18 16:25:07 +02:00
Tim-Philipp Müller
6347ec522d rtpjp2kpay: pre-allocate buffer-list of the right size 2014-06-18 14:54:59 +01:00
Tim-Philipp Müller
ccb7380689 rtpjpegpay: pre-allocate buffer list of the right size 2014-06-18 14:54:59 +01:00
Tim-Philipp Müller
70bfc35756 rtpmp4vpay: pre-allocate buffer list of the right size 2014-06-18 14:54:59 +01:00
Tim-Philipp Müller
4b1f771e4d rtpvp8pay: allocate bitreader on the stack 2014-06-18 14:54:59 +01:00
Tim-Philipp Müller
725b8f272b rtpvp8pay: post error message on bus on error and don't use g_message() 2014-06-18 14:54:59 +01:00
Tim-Philipp Müller
f4db7443ae rtpvp8pay: couple of minor optimisations
Pre-allocate buffer list of the right size to avoid re-allocs.
Avoid plenty of double runtime cast checks and re-doing the
same calculation over and over again in rtp_vp8_calc_payload_len().
Only call gst_buffer_get_size() once.
2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
6c9e2194d2 rtpgstpay: pre-allocate buffer list of the right size
To avoid re-allocs.
2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
01ee993d8d rtph264pay: pre-allocate bufferlist of the right size
To avoid unnecessary re-allocs.
2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
c7c72c00b1 rtph264pay: push single buffer directly, no need to wrap it in a bufferlist
No point in a buffer list if we just have one single
buffer to push. Fix up unit test to handle that case
as well.
2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
0f5da64de3 rtpvrawpay: make chunks per frame configurable
Bit of a misnomer because it's really chunks per field
and not per frame, but we're going to ignore that for
the time being.
2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
2cf13b603f rtpvrawpay: remove unused variables 2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
a09e237b85 rtpvrawpay: pre-allocate buffer lists of sufficient size
Avoids unnecessary reallocs when appending buffers
to the bufferlist.
2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
15a33ccc65 rtpvrawpay: micro-optimise variable access in inner loop
Store some values that don't change during the execution
of the inner loops locally, so the compiler knows that too.
2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
fdf95fecbd rtpvrawpay: use buffer lists
Collect buffers to send out in buffer lists instead of
pushing out single buffers one at a time. For HD video
each frame might easily add up to a couple of thousand
packets, multiply that by the frame rate and that's a
lot of push() and sendmsg() calls per second.

A good reason to push out buffers as early as possible is
latency, so we don't accumulate the whole frame in a single
buffer list, but instead push it out in a few chunks, which
is hopefully a reasonable compromise.
2014-06-18 14:54:58 +01:00
Tim-Philipp Müller
054f774455 rtptheoradepay: fix double frees
Fix double-frees introduced to fix another coverity report.

CID 1223053
2014-06-16 12:03:38 +01:00
Vincent Penquerc'h
25c26a4c4c rtptheordepay: fix leaks
Coverity 1212163
2014-06-12 11:24:15 +01:00
Vincent Penquerc'h
8e80478cf7 rtpg729pay: leak fixes
Coverity 1212159
2014-06-12 11:16:08 +01:00
Vincent Penquerc'h
fe4c5b92b1 rtph263pay: fix leak
Coverity 1212157
2014-06-12 11:11:38 +01:00
Vincent Penquerc'h
6ef26e4a8a rtph263pay: fix leaks
Coverity 1212149
2014-06-12 10:43:53 +01:00
Vincent Penquerc'h
c58a2d9bbb rtpdvpay: catch failures to map buffer
Coverity 1139741
2014-06-12 10:31:47 +01:00
Wim Taymans
a5a7649831 h264depay: make sure we call handle_nal for each NAL
Call handle_nal for each NAL in the STAP-A RTP packet. This makes
sure we correctly extract the SPS and PPS.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730999
2014-05-30 16:51:37 +02:00
Guillaume Desmottes
d089f99a39 rtp/README: update pipelines to work with 1.0
- Use gst-libav encoders/decoders instead of gst-ffmpeg
- gstrtpjitterbuffer -> rtpjitterbuffer
- gst-launch-0.10 -> gst-launch-1.0
- Add 'videoconvert' element
- xvimagesink -> autovideosink

https://bugzilla.gnome.org/show_bug.cgi?id=729247
2014-05-05 20:23:56 -04:00