Commit graph

1178 commits

Author SHA1 Message Date
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
Vincent Penquerc'h
60ba2d7aee rtpqdmdepay: remove pointless check
Besides, the pointer was dereferenced earlier anyway.

Coverity 1139853
2014-05-02 14:09:02 +01:00
Vincent Penquerc'h
a55b8e9c00 rtpvrawpay: guard against pathological "no space" condition
Even if one woul hope one pixel can fit in a MTU, ensure we do not
overwrite a buffer if this is not the case.

Spotted while looking at Coverity 1208786
2014-04-30 14:50:44 +01:00
Vincent Penquerc'h
dfa2df1c88 rtpjpegdepay: sanity check for NULL qtable
Can happen (at least in crafted stream)

Coverity 1208778
2014-04-30 11:52:10 +01:00
Wim Taymans
9994ff2c6c rtpvraw: use plane pointers when needed
Pack/unpack planar formats to/from the first plane.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729058
2014-04-28 14:45:57 +02:00
Tim-Philipp Müller
c9597298f9 docs: remove outdated and pointless 'Last reviewed' lines from docs
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.
2014-04-26 23:35:17 +01:00
Göran Jönsson
80967c7638 gstrtph264pay: Reset sps pps variable when state change.
Reset last_spspps and sps/pps arrays  when state transition
GST_STATE_CHANGE_PAUSED_TO_READY.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726015
2014-04-21 12:07:20 +02:00
Josep Torra
eaee14aff4 rtph264depay: only guess AU boundaries when aren't indicated by marker
The marker bit isn't mandatory and we had in place code to guess AU
boundaries by detecting a new picture start. This guessing code
didn't work with interlaced content that has proper marker bits
to indicate the AU boundaries. It was leaking the first field buffer
and producing a corrupted output.

fixes: https://bugzilla.gnome.org/show_bug.cgi?id=728041
2014-04-12 04:42:36 +02:00
Ognyan Tonchev
2143a6e452 jpegpay: consider header len when calculating payload len
Fixed https://bugzilla.gnome.org/show_bug.cgi?id=726777
2014-03-27 09:45:20 +01:00
Olivier Crête
15d276058e rtp: Remove caps restrictions from RTP depayloader sink caps
Remove caps restrictions that correspond to the default and are not
required in SDP. With the new usage of having pads require a subset
of the caps, they will make the negotiation fail.
2014-03-06 12:06:43 -05:00
Olivier Crête
5a9b988b85 rtpspeexdepay: Remove caps restrictions for depayloader
The "encoding-params" is optional in the SDP, because we now require
a subset of the caps, it would fail caps negotiatioin if it wasn't present.
So removed it from the template caps.
2014-03-06 11:03:04 -05:00
Göran Jönsson
53ffd9e1ca rtph264pay: only update last_spspps time if all sps/pps got sent successfully
This fixes an issue with gst-rtsp-server where no sps and pps are
sent for the first intra frame, because the payloader starts working
already when receiving DESCRIBE but there is no transports so it tries
to send sps and pps, but that fails with a FLUSHING flow. But the time
for last sent sps and pps would still be set, so when PLAY arrives and
the first intra frame is to be sent there is no sps and pps sent due to
that time since last sps pps is less than spspps_interval.

https://bugzilla.gnome.org/show_bug.cgi?id=724213
2014-02-25 10:48:24 +00:00
Sebastian Dröge
3d8f078b61 rtpac3depay: Remove unused variable 2014-02-08 17:21:19 +01:00
Sebastian Dröge
e178cf60ae rtpvp8pay: Don't leak input buffers
https://bugzilla.gnome.org/show_bug.cgi?id=722414
2014-01-20 10:13:19 +01:00
Justin Joy
70be4fa24a rtpg726pay: don't leak encoding_name string
https://bugzilla.gnome.org/show_bug.cgi?id=722159
2014-01-14 10:29:47 +01:00
Edward Hervey
91c5b09fb4 rtpvrawpay: Add missing break
COVERITY CID 1139762
2013-12-30 17:20:37 +01:00
Edward Hervey
ac40045d0d rtpqcelpdepay: Remove uneeded variable 2013-12-30 13:50:35 +01:00
Sebastian Dröge
d585bd7bbd rtptheorapay: Don't send headers twice if we got them from the caps already 2013-12-04 21:58:29 +01:00
Sebastian Dröge
d105de6e0f rtptheorapay: Don't leak config data when receiving a second CAPS event 2013-12-04 21:58:29 +01:00
Sebastian Dröge
0915d696c7 rtpvorbispay: Don't send headers twice if we got them from the caps already 2013-12-04 21:58:29 +01:00
Sebastian Dröge
967280df42 rtpvorbispay: Don't leak config data when receiving a second CAPS event 2013-12-04 21:58:29 +01:00
Sebastian Dröge
d87f6cf483 rtpstreamdepay: Add RFC4571 RTP stream depayloading element
https://bugzilla.gnome.org/show_bug.cgi?id=719829
2013-12-04 21:58:29 +01:00
Sebastian Dröge
c5284dc047 rtpstreampay: Add RFC4571 RTP stream payloading element
https://bugzilla.gnome.org/show_bug.cgi?id=719829
2013-12-04 21:58:29 +01:00
Jonas Holmberg
0ab0421759 rtph264pay: Map inbuffer once only
Do not call gst_buffer_extract() twice since each call will map and
unmap the biffer.

https://bugzilla.gnome.org/show_bug.cgi?id=719434
2013-11-28 16:08:40 -05:00
Wim Taymans
3a1199c2f7 rtpvorbisdepay: handle packets > 0xffff
Handle input packet sizes larger than 16 bits in the depayloader.
Remove size restrictions on the payloader.
2013-11-21 11:32:15 +01:00
Wim Taymans
43e9b56122 rtptheoradepay: handle packets > 0xffff
Reorganize some things in the depayloader so that it can handle packets larger
than 16 bits.
Remove the size restriction on the payloader.
2013-11-21 11:30:28 +01:00
Wim Taymans
a065b4fcde gstpay: only send one caps
Only send one caps in a packet. Two caps can happen when setcaps is called and
the config-interval expires at the same time.
2013-11-13 12:02:57 +01:00
Sebastian Dröge
111982de28 rtpvp8pay: Make Picture ID mode configurable and default to no picture ID
Some implementations (linphone) only support no picture at all in the
stream and will fail if one is provided.

https://bugzilla.gnome.org/show_bug.cgi?id=711497
2013-11-05 17:26:49 +01:00
Paul HENRYS
8eceb8f327 Add call to gst_rtp_h264_pay_clear_sps_pps() when receiving a STREAM_START event
https://bugzilla.gnome.org/show_bug.cgi?id=692787
2013-11-04 14:36:28 -05:00
Ognyan Tonchev
93d5e182d2 rtpgstpay: Fix memory leak
We were leaking the GList nodes of the pending buffers.

https://bugzilla.gnome.org/show_bug.cgi?id=709079
2013-10-02 11:07:16 +02:00
Jan Schmidt
299d3f5c42 rtp: Remove bogus extra caps from L24 template.
The extra caps entry in the template was making it sometimes
get plugged for any dynamically allocated payload type.
2013-09-13 23:27:49 +10:00
Rico Tzschichholz
8ed1ff6821 rtp: Add missing headers tp fix make dist
In addition to a956a6ceb2
2013-09-13 14:06:13 +02:00
David Holroyd
a956a6ceb2 rtp: add L24 pay and depayloader
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707734
2013-09-09 15:13:46 +02:00
Wim Taymans
d851b8a8b4 rtpmpvpay: Fix RTP buffer allocation in rtpmpvpay
RTP buffer allocation should not be done with padding for the specific MPEG2
header as the padding is done at the end of the buffer and the last byte is
the size of the padding.

https://bugzilla.gnome.org/show_bug.cgi?id=706970
2013-08-29 13:15:15 +02:00
Wim Taymans
f626e29897 jpegdepay: add some more debug 2013-08-21 12:56:35 +02:00
Wim Taymans
77ed44a88a rtpgstdepay: only push events when they changed
Keep track of the STREAM_START and TAG events and only push them
when they changed.
2013-08-21 12:10:00 +02:00
Wim Taymans
b144809b7c rtpgstpay: taglists should not be merged in 1.0 2013-08-21 10:52:59 +02:00
Wim Taymans
69b0dcd7df rtpgstdepay: flush on FLUSH_STOP event 2013-08-21 10:28:50 +02:00
Wim Taymans
5ff9093843 rtpgstpay: reset on state change
Do full reset on state change to READY
2013-08-21 10:03:52 +02:00
Wim Taymans
ae9239aac7 rtpgstpay: reset on FLUSH_STOP
Clear the adapter and pending buffer list on FLUSH_STOP.
2013-08-21 09:55:20 +02:00
Wim Taymans
2e8955df39 rtpgstpay: don't use clock for config interval
We can't use the clock to time our config-interval because we are not
live (or there might not be a clock or the clock might not be running).
Instead just simply take the timestamp diff.
2013-08-21 09:39:30 +02:00
Wim Taymans
182f96ff79 rtpgstay: don't use // comments 2013-08-21 09:33:04 +02:00
Youness Alaoui
62a6f58697 Send a stream-start whenever we send tags
This is to make sure tags are cleared on the client if the
stream-start was previously lost, otherwise, the client may end
up with a merged taglist of multiple songs
2013-08-21 09:06:01 +02:00
Youness Alaoui
05bcfee5a3 rtpgstpay: Add a config-interval property to resend the caps/tags at a regular interval
This is useful in case the packet containing the inlined caps was lost
or if new client joins an already running RTP stream and they missed
the previous tag events.
This also makes the payloader keep a list of merged tags so the retransmitted
tag event contains all previously received. A STREAM_START event will
flush the list of tags.
2013-08-21 09:06:01 +02:00
Youness Alaoui
1f4ca28868 rtpgstpay: Refactor the setcaps and use new method to send arbitrary caps at any time 2013-08-21 09:06:01 +02:00
Youness Alaoui
9257409613 rtpgstpay: Do not flush events for stream-start and avoid conflict between event and pending inline caps 2013-08-21 09:06:01 +02:00
Youness Alaoui
2d53289b6b rtpgstpay: Add a create_from_adapter API and use a list of GstBufferList
This is necessary to fix event/caps sending. If we send a STREAM_START
packet, it will cause an error because the stream didn't receive its
caps and new-segment events, so we must wait for the first buffer before
sending the stream-start event buffer. However, the caps will be sent
at the same time and so the 'inline caps' will be set for the event.
We need to be able to payload individual packets (data, caps or events)
and only send them when we call flush.
2013-08-21 09:06:01 +02:00
Youness Alaoui
0070ba76f2 rtpgstpay: Add etype=4 for payloading GST_EVENT_STREAM_START 2013-08-21 09:06:01 +02:00
Youness Alaoui
6155b27971 rtpgstpay: Fix typo, GST_EVENT_CUSTOM_BOTH has etype of 3 2013-08-21 09:06:01 +02:00
Wim Taymans
db90f6e68d h264depay: init debug category early
Init the debug variable when we register the element because it is also used by
the payloader element when it calls the add_sps_pps method.
2013-08-16 17:12:19 +02:00
Olivier Crête
4c6e636720 rtph264pay: Use the SPS/PPS handling function from the depayloader
Remove duplicated copies

https://bugzilla.gnome.org/show_bug.cgi?id=705553
2013-08-13 10:38:23 -04:00
Olivier Crête
742b90747d rtph264depay: Make the SPS/PPS deduplication function generic
Make it not touch any internals of the depayloader

https://bugzilla.gnome.org/show_bug.cgi?id=705553
2013-08-13 10:38:23 -04:00
Tim-Philipp Müller
957c8e3e61 rtpvp8depay: mark key frames and delta frames properly
https://bugzilla.gnome.org/show_bug.cgi?id=705550
2013-08-07 11:14:38 +01:00
Wim Taymans
cc92ef1db2 vrawdepay: fix UYVP format 2013-07-24 12:42:31 +02:00
Wim Taymans
8191b6fcd2 vrawpay: fix UYVP format 2013-07-24 12:41:58 +02:00
Wim Taymans
37af93c361 vrawpay: fix caps 2013-07-24 12:41:44 +02:00
Olivier Crête
54c5a7f690 rtp: Use gst_adapter_take_buffer_fast() where possible in RTP payloaders 2013-07-16 15:37:49 -04:00
Sebastian Dröge
0cc77d8e30 rtph263ppay: Don't pass upstream filter caps to downstream
Downstream usually can't accept video/x-h263 but only application/x-rtp,
so we would always get an empty intersection here.

https://bugzilla.gnome.org/show_bug.cgi?id=702632
2013-07-08 14:10:44 +02:00
J. Rick Ramstetter
f01b751e52 rtp: Fix documentation and comments to use rtpbin instead of old gstrtpbin
https://bugzilla.gnome.org/show_bug.cgi?id=703426
2013-07-02 10:12:17 +02:00
Wim Taymans
c469434ea8 vorbispay: add support for config-interval
Align code with the theora payloader and add support for the config-interval to
periodically send out the config headers.
2013-06-28 15:21:56 +02:00
Wim Taymans
006562c9f4 theorapay: small cleanups 2013-06-28 15:21:12 +02:00
Wim Taymans
cdc66462ce theorapay: handle streamheaders as well 2013-06-28 12:08:19 +02:00
Wim Taymans
3169432ed4 vorbispay: always collect headers on data
When we see a data packet, always check if we need to collect any previous
headers.
2013-06-28 12:07:58 +02:00
Wim Taymans
6c716dfc25 vorbispay: handle streamheader as well
Take config strings from the streamheader when we can

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=664312
2013-06-28 11:43:17 +02:00
David Svensson Fors
692206d3a7 rtph264pay: avoid double buffer unmap on error
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703171
2013-06-27 17:14:11 +02:00
Wim Taymans
4258ddcc36 jpegpay: turn some errors into warnings
Turn some errors into warnings, we can continue processing so this should
not be fatal.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=657079
2013-06-26 20:49:41 +02:00
Jens Georg
745be945ce rtpmp2tdepay: accept mislabelled streams from GStreamer 0.10 as well
The mp2t payloader in 0.10 mislabelled the streams as MP2T-ES
instead of MP2T, so accept that as well for compatibility reasons.

https://bugzilla.gnome.org/show_bug.cgi?id=702457
2013-06-17 15:39:17 +01:00
Wim Taymans
1f0600ee6f Revert "rtph264pay: Restructuring to allow for adding optional caps"
This reverts commit 61666898cf.

This commit changes what the set_sps_pps() function does, not it doesn't
set caps anymore (and should have been renamed). The main problem is that
not all call sites are updated and thus leak the string.
2013-05-31 15:18:48 +02:00
Wim Taymans
1516c14881 Revert "rtph264pay/depay: Add frame dimensions a payloaded caps"
This reverts commit 3dca756a5d.

The H264 RTP spec has no attributes for width and height.
2013-05-31 15:11:12 +02:00
Wim Taymans
b79d217396 Revert "rtph264pay/depay: Add optional framerate caps for use in SDP"
This reverts commit d8825e2a5c.

There is no framerate attribute in the h264 RTP spec.
2013-05-31 15:09:51 +02:00
Wim Taymans
190b3d6688 Revert "rtpjpegpay/depay: Replace framesize caps with width/height"
This reverts commit 0075d111b4.

Extra application/x-rtp are SDP fields, which are strings.
2013-05-31 15:08:16 +02:00
Wim Taymans
f870cef8bc Revert "rtpjpegpay/depay: Replace framerate caps field with fraction"
This reverts commit 9fd25a810b.

We deal with sdp attributes in application/sdp, which are always strings.
2013-05-31 15:05:51 +02:00
Sebastian Rasmussen
9fd25a810b rtpjpegpay/depay: Replace framerate caps field with fraction
The previous implementation had the formatting of SDP attributes happen
in each RTP payloader, now instead the constituent values are propagated
as caps fields. This allows for applications to do SDP offer/answer
based on caps negotiation.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748
2013-05-23 21:05:49 +02:00
Sebastian Rasmussen
0075d111b4 rtpjpegpay/depay: Replace framesize caps with width/height
The previous implementation had the formatting of SDP attributes happen
in each RTP payloader, now instead the constituent values are propagated
as caps fields. This allows for applications to do SDP offer/answer
based on caps negotiation.

Keep parsing a-framerate, x-framerate and x-dimensions in rtpjpegdepay
to be backwards compatible with previous payloaders.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748
2013-05-23 21:05:43 +02:00
Sebastian Rasmussen
d8825e2a5c rtph264pay/depay: Add optional framerate caps for use in SDP
This allows for applications to format SDP attributes and still do SDP
offer/answer based on caps negotiation.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700749
2013-05-23 21:04:17 +02:00
Sebastian Rasmussen
3dca756a5d rtph264pay/depay: Add frame dimensions a payloaded caps
This allows for applications to format SDP attributes and still do SDP
offer/answer based on caps negotiation.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700749
2013-05-23 21:04:11 +02:00
Sebastian Rasmussen
61666898cf rtph264pay: Restructuring to allow for adding optional caps
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700749
2013-05-23 21:04:00 +02:00
Sebastian Rasmussen
2361567bae rtpjpegpay/depay: Add framesize caps for use in SDP
The format of the value adheres to RFC6064 and it is meant to be parsed
and included in the SDP sent by gst-rtsp-server to its clients.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748
2013-05-21 09:09:03 +02:00
Sebastian Rasmussen
919eed0787 rtpjpegpay: Add optional framerate caps for use in SDP
The format of the value adheres to RFC4566 and it is meant to be parsed
and included in the SDP sent by gst-rtsp-server to its clients.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748
2013-05-21 09:08:21 +02:00
Michael Olbrich
d1c56376d6 rtpmp4apay: clear config buffer before using it
This is necessary because parts of the memory are only modified with "|="

https://bugzilla.gnome.org/show_bug.cgi?id=700514
2013-05-18 10:57:56 +01:00
Sebastian Dröge
f28ab45f3e rtpgstpay: First let baseclass handle events, then put them into the stream
Fixes handling of sticky events.

https://bugzilla.gnome.org/show_bug.cgi?id=700213
2013-05-13 13:44:35 +02:00
Sebastian Rasmussen
9532b04947 rtpgstpay: fix invalid memory access in event handler
First process event in payloader, then hand it to the
base class which takes ownership of the event.

https://bugzilla.gnome.org/show_bug.cgi?id=699637
2013-05-04 10:49:23 +01:00
Andoni Morales Alastruey
4a78a77e65 rtp: fix duplicated symbols with libvpx 2013-05-02 14:03:33 +02:00
Sebastian Dröge
ae05ed4f05 rtph264pay: If the adapter is empty on EOS don't try to map its content
https://bugzilla.gnome.org/show_bug.cgi?id=699314
2013-05-01 15:49:45 +02:00
Wim Taymans
1df2e623b5 docs: add some pay/depayloaders
See https://bugzilla.gnome.org/show_bug.cgi?id=551631
2013-04-25 14:05:55 +02:00
Wim Taymans
5ba3fd3c63 vrawdepay: return output buffer from process
Return the output buffer from the process function instead of pushing
it ourselves. This way, the subclass can actually deal with the return
value of the push.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=693727
2013-04-24 16:24:25 +02:00
Wim Taymans
eac9efb92e rtp: a marker bit should translate to RESYNC
A marker bit on an audio packet does not mean a DISCONT (in the GStreamer sense
of missing data) but it means that the packet is the end of a talkspurt and thus
a good opportunity to resync to the clock. Use the RESYNC buffer flag to note
this.
Real discontinuities are marked with DISCONT still when the seqnum has a GAP or
when the input buffer has the DISCONT flag set.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=627204
2013-04-24 15:42:45 +02:00
Sebastian Dröge
fdb667ae00 rtpjpegdepay: Drop frame if it's less than 2 bytes large
https://bugzilla.gnome.org/show_bug.cgi?id=677560
2013-04-22 10:19:29 +02:00
Sebastian Dröge
b0b0557c48 gst: Add better support for static plugins 2013-04-15 15:54:11 +02:00
Wim Taymans
9d7519f66e rtp: register tag image types
The rtpgstdepay needs the type to be available in order to deserialize the
event.
2013-04-12 16:18:42 +01:00
Wim Taymans
b1f4587d75 rtpgstdepay: handle event parse failures better 2013-04-12 16:18:42 +01:00
Andreas Fenkart
20d3ec8810 rtpsbcdepay: fix sbc frame length calculation for mono and stereo modes
https://bugzilla.gnome.org/show_bug.cgi?id=697463
2013-04-09 23:17:57 +01:00
Wim Taymans
91a3afc4dc gstpay: use bufferlist to avoid memcpy 2013-04-09 16:53:31 +02:00
Nicola Murino
c41c16424d rtpsbcdepay: fix printf format compiler warnings
https://bugzilla.gnome.org/show_bug.cgi?id=697343
2013-04-05 13:50:19 +01:00
Olivier Crête
f8831c0cd2 rtpsbcdepay: Rank as secondary
This way, it will be selected by decodebin
Bug reported by andreas.fenkart@streamunlimited.com

https://bugzilla.gnome.org/show_bug.cgi?id=697227
2013-04-03 18:25:36 -04:00
Wim Taymans
ac2bcfa833 theorapay: add delta-unit to output frames 2013-03-31 19:14:04 +02:00
Josep Torra
509631f60b rtp: fixes debug message printf related compiler warnings in SBC depayloader 2013-03-30 09:44:41 +01:00
Arun Raghavan
87bdad4bfc rtp: Add an rtpsbcdepay element
Pretty straightforward - takes SBC encapsulated in RTP, depayloads, and
pushes out SBC buffers.

https://bugzilla.gnome.org/show_bug.cgi?id=690582
2013-03-28 17:22:33 +00:00
Tim-Philipp Müller
477cc51fe7 rtp: fix SBC payloader
Init RTP buffer on stack correctly, so mapping it works
without criticals and the payloader actually works.
2013-03-27 22:18:34 +00:00
Ognyan Tonchev
3f8ad30cee rtph264pay: Don't use upstream caps with peer_query_caps ()
Calling gst_pad_peer_query_caps () on the src pad with the caps
upstream can produce as a filter from gst_rtp_h264_pay_getcaps ()
is wrong and makes caps negotiation fail if upstream caps are not
NULL.

https://bugzilla.gnome.org/show_bug.cgi?id=695629
2013-03-11 16:55:13 -04:00
Olivier Crête
df5ca83baf rtpmp4gdepay: streamtype is not put by all RTSP server, not make it optional
Specific case here is Wowza 3.5.0
2013-02-26 14:19:10 -05:00
Sebastian Dröge
a7ddbc03fe rtp-payloading: Fix unit test caps and AMR depayloader sink template caps
Fields were missing from the actual caps, or too many fields
existed in the template caps.
2013-02-13 12:02:46 +01:00
Wim Taymans
4397c8ffbf rtpdepay: remove payload type restrictions
Remove the pt restrictions for all the depayloaders that have an
encoding-name. We can use this to autoplug decoders.
Remove the encoding-name for all the payloaders with a fixed payload
type.
We now either have an encoding-name or a pt in the sinkpad caps of
a depayloader.

See https://bugzilla.gnome.org/show_bug.cgi?id=639292
2013-01-28 12:41:04 +01:00
Marc Leeman
bab2f3c92b rtp: remove payload requirements from selected depayloaders
encoding name is required in the caps and is a better fit for autoplugging than
the pt value. Hardware manufacturers have a bad habit of skimming through RFCs
and in this case; use unassigned numbers for encoders instead of dynamic
numbers.

In essence, this patch will add support for a lot of Bosch hardware encoders
without breaking autoplugging.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=639292
2013-01-28 12:23:41 +01:00
Tim-Philipp Müller
9455a3aee1 rtpsbcpay: update some fields in the caps to their new name
and to match the parser. "mode" got renamed to "channel-mode"
and "allocation" to "allocation-method".
2013-01-16 10:19:36 +00:00
Tim-Philipp Müller
39ef892938 rtp: import rtpsbcpay from bluez and port to 1.0
Compiles, but not tested yet (sbc elements still need to be ported).

https://bugzilla.gnome.org/show_bug.cgi?id=690582
2013-01-10 12:43:50 +00:00
Marcel Holtmann
4196feb659 rtpsbcpay: Remove workaround for compiler warnings 2013-01-10 00:18:03 +00:00
Marcel Holtmann
fe79c60d74 rtpsbcpay: Add pragma based workaround for GStreamer warnings 2013-01-10 00:18:03 +00:00
Marcel Holtmann
08e95e7249 rtpsbcpay: Update copyright information 2013-01-10 00:15:36 +00:00
Marcel Holtmann
7fa03c0076 rtpsbcpay: Fix signed/unsigned comparison issue within GStreamer plugin 2013-01-10 00:15:35 +00:00
Marcel Holtmann
27a6b0abfe rtpsbcpay: Update copyright information 2013-01-10 00:15:35 +00:00
Marcel Holtmann
f890079aae rtpsbcpay: First attempt in fixing compiler warnings (still needs cleanup) 2013-01-10 00:15:35 +00:00
Johan Hedberg
7d4f846112 rtpsbcpay: More coding style fixes 2013-01-10 00:15:35 +00:00
Luiz Augusto von Dentz
151ad9b28d rtpsbcpay: Remove possible extra memcpy for gstreamer plugin. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
69c8374b7c rtpsbcpay: Fix bug sending empty packages and remove a buffer copy. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
7b3e4356ea rtpsbcpay: Fix runtime warnings of gstreamer plugin. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
f74f061f3b rtpsbcpay: Update gstreamer plugin to use new sbc API. 2013-01-10 00:13:14 +00:00
Marcel Holtmann
b9be04f07b rtpsbcpay: Update copyright information 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
687400ecf4 rtpsbcpay: Fixes gstreamer caps and code cleanup. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
a4f9624261 rtpsbcpay: Fix gtreamer payloader sending fragmented frames. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
41e2f4f544 rtpsbcpay: Fix use of gstreamer plugin with rhythmbox and banshee and rtp timestamps. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
96971cd323 rtpsbcpay: Make a2dpsink to act like a bin and split the payloader. 2013-01-10 00:13:14 +00:00
Jonas Holmberg
e12457f138 rtpjpegpay: handle width and height > 2040
If width or height is greater than 2040 set width and height to zero in
the rtp header and add x-dimensions to outcaps.

Solves #684955
2012-12-20 15:40:49 +01:00
Thijs Vermeir
de41376231 rtp: use appropriate printf format for gsize 2012-12-18 16:02:09 +01:00
Tim-Philipp Müller
bdf3c77828 gst_adapter_prev_timestamp -> gst_adapter_prev_pts
https://bugzilla.gnome.org/show_bug.cgi?id=675598
2012-11-14 00:13:36 +00:00
Christian Fredrik Kalager Schaller
485505f323 Fix vp8rtp header names in Makefile 2012-11-07 13:36:33 +01:00
Tim-Philipp Müller
230cf41cc9 Fix FSF address
https://bugzilla.gnome.org/show_bug.cgi?id=687520
2012-11-04 00:07:18 +00:00
Wim Taymans
9857e6af4d vrawdepay: don't access rtp buffer after unmap
Read the marker bit before we unmap the rtp packet.
2012-11-02 18:48:17 +00:00
Tim-Philipp Müller
5ac789408b rtpvp8: include config.h and minor style fixes 2012-11-01 21:10:21 +00:00
Tim-Philipp Müller
4a849d6690 rtp: fix tabs/space mess in Makefile.am 2012-11-01 20:53:48 +00:00
Tim-Philipp Müller
321acd14dc rtp: move VP8 payloader and depayloader from -bad
Spec is still in draft state, but should hopefully not
change much now. Besides, we announce things as VP8-DRAFT-IETF-01
in our caps, so even if things change in incompatible ways it
should not break anything.

https://bugzilla.gnome.org/show_bug.cgi?id=687263
2012-11-01 20:53:48 +00:00
Tim-Philipp Müller
44efab8e3d rtpvp8: use gst_element_class_set_static_metadata()
where possible. Avoids some string copies. Also re-indent
some stuff. Also some indent fixes here and there.
2012-11-01 20:53:48 +00:00
Mark Nauwelaerts
bc7dbbbd4f rtpvp8: replace gst_element_class_set_details_simple with gst_element_class_set_metadata 2012-11-01 20:53:48 +00:00
Sebastian Dröge
4853001547 rtpvp8: update for GST_PLUGIN_DEFINE() API changes 2012-11-01 20:53:48 +00:00
Wim Taymans
fccfca38d4 rtpvp8: update for buffer changes 2012-11-01 20:53:48 +00:00
Danilo Cesar Lemes de Paula
3edffb13e3 rtpvp8; fix compatibility with the third draft
https://bugzilla.gnome.org/show_bug.cgi?id=671073
2012-11-01 20:53:48 +00:00
Mark Nauwelaerts
d9581832a0 rtpvp8: port some more to new memory API 2012-11-01 20:53:47 +00:00
Olivier Crête
c6761daa27 rtpvp8: port to 0.11 2012-11-01 20:53:47 +00:00
Sebastian Dröge
2c5ea76bdc rtpvp8pay: Fix typo 2012-11-01 20:53:47 +00:00
Youness Alaoui
1cf155d70d rtpvp8: Update the pay/depay to the ietf-draft-01 spec 2012-11-01 20:53:47 +00:00
Vincent Penquerc'h
88aade4150 rtpvp8: fix bitstream parsing using the wrong kind of bitreader
VP8 uses a probabilistic bool coder, not a straight bit coder.
This fixes parsing when error-resilient is set.

This commit includes a copy of libvpx's bool coder, BSD licensed.

https://bugzilla.gnome.org/show_bug.cgi?id=652694
2012-11-01 20:53:47 +00:00
Olivier Crête
97c3f3617c rtpvp8: Reject unknown bitstream versions 2012-11-01 20:53:47 +00:00
Edward Hervey
74a1a704bf rtpvp8: Fix unitialized variable
Makes macosx compiler happy.
2012-11-01 20:53:47 +00:00
Sjoerd Simons
6ed6318076 rtpvp8depay: Accept packets with only one byte of data
When fragmenting partions it can happen that an RTP packet only caries 1
byte of RTP data.
2012-11-01 20:53:47 +00:00
Sjoerd Simons
a45e7a3fc0 rtpvp8pay: Treat the frame header just like any other partition
When setting up the initial mapping just act as if the global frame
information is another partition. This saves special-casing it later in
the actual packetizing code.
2012-11-01 20:53:47 +00:00
Sjoerd Simons
e9f4e9342f rtpvp8: Add simple payloaders and depayloaders for VP8
Minimal implementation of http://www.webmproject.org/code/specs/rtp/,
version 0.3.2
2012-11-01 20:53:47 +00:00
Wim Taymans
d6fd0ebd04 gstpay: fix for 1.0 events
Caps events are sometimes not followed by a buffer but by an event. Flush any
pending caps before we make a packet with the event.
Chain up to the parent event handler before we attempt to push RTP packets, it
might be a segment event.
2012-11-01 18:42:39 +00:00
Wim Taymans
05232c55a5 gstdepay: fix small leak 2012-11-01 18:42:24 +00:00
Wim Taymans
08e5a197b4 gstdepay: add support for events
Conflicts:
	gst/rtp/gstrtpgstdepay.c
2012-11-01 18:18:19 +00:00
Wim Taymans
54b783b5a3 rtpgstpay: add support for sending events
We currently only send tags and custom events. The other events
might interfere with the receiver timings or are otherwise handled
by RTP.

Conflicts:
	gst/rtp/gstrtpgstpay.c
2012-11-01 18:06:11 +00:00
Wim Taymans
6502d08e43 gstpay: rewrite payloader
Use adapter to assemble the payload and make a flush function to
turn this payload into (fragmented) packets.

Conflicts:
	gst/rtp/gstrtpgstpay.c
	gst/rtp/gstrtpgstpay.h
2012-11-01 17:57:52 +00:00
Wim Taymans
c0713e4b80 gstdepay: check for correct fragment offset
Make sure we only insert the rtp packet in the adapter when the
frag_offset matches. When the first packet of a fragment is dropped,
it avoids putting the remaining packets in the adapter and processing
the partial fragment.

Conflicts:
	gst/rtp/gstrtpgstdepay.c
2012-11-01 12:09:47 +00:00
Wim Taymans
8a402e0c06 gstpay: set C flag on all buffers of the fragment
Set the C flags on all the fragments instead of only those with
caps in them. This makes it easier in the receiver to check if there
is a caps in the assembled fragments just by looking at the last RTP
packet flags.
2012-11-01 12:06:08 +00:00
Wim Taymans
d78ff07f7d gstdepay: use the capsversion
Take the caps from the input caps and store it in the slot given
by capsversion.
2012-11-01 11:37:44 +00:00
Wim Taymans
936c3819b5 gstpay: send caps inline
Place the capsversion on the outgoing caps so that they end up in
an SDP as well. Receivers need to know what capsversion a particular
caps is for to be able to match the caps to the CV in the RTP packets.
Place the caps inside the RTP packet whenever the caps change.

Based on patch by Andrzej Bieniek <andrzej.bieniek@pure.com>

Conflicts:
	gst/rtp/gstrtpgstpay.c
	gst/rtp/gstrtpgstpay.h
2012-11-01 11:34:33 +00:00
Andrzej Bieniek
3b1931a039 gstpay: add debug
Conflicts:
	gst/rtp/gstrtpgstpay.c
2012-11-01 11:28:50 +00:00
Andrzej Bieniek
ee5ecc7773 depay: correctly skip caps header size
Conflicts:
	gst/rtp/gstrtpgstdepay.c
2012-11-01 11:27:13 +00:00
Wim Taymans
e9040e90a5 jpegdepay: store quant tables in zigzag order 2012-10-17 14:23:01 +02:00
Rasmus Rohde
47a8eb7ca8 gstrtpdepay: don't leak input buffer
The rtp buffer is never unmapped in the normal code exit path
of gst_rtp_gst_depay_process(..) resulting in a memory leak.

https://bugzilla.gnome.org/show_bug.cgi?id=685512
2012-10-04 19:44:28 +01:00
Patricia Muscalu
7a863e4d8d rtph264pay: do not push unmapped data
Also do not use a GstBuffer after it has been pushed into the adapter.

https://bugzilla.gnome.org/show_bug.cgi?id=685213
2012-10-04 09:22:50 +01:00
Wim Taymans
dbe941338d rtpvrawdepay: negotiate pool with srcpad caps 2012-09-27 14:15:50 +02:00
Olivier Crête
bc252d29ee rtph264pay: Make sure the caps don't have duplicated sps/pps 2012-09-21 17:36:12 -04:00
Wim Taymans
829c80ce6c fix more caps 2012-09-14 13:30:37 +02:00
Mark Nauwelaerts
8d93246b93 gst: adjust comment style 2012-09-10 14:31:02 +02:00
Mark Nauwelaerts
f24b58d19c rtpamrdepay: unmap rtp buffer
... thereby plugging a memleak.
2012-09-07 15:25:53 +02:00
Mark Nauwelaerts
fa90dfc4df rtph264pay: avoid crashing on NULL access in debug message 2012-09-07 15:25:52 +02:00
Mark Nauwelaerts
8f4bfeb698 rtph263ppay: plug caps leak 2012-09-07 15:25:52 +02:00
Tim-Philipp Müller
9bf90f47cf video/x-xvid -> video/mpeg,mpegversion=4 2012-09-03 02:51:24 +01:00
Tim-Philipp Müller
4bb52bbadf docs: gst-launch -> gst-launch-1.0 and ffmpegcolorspace -> videoconvert 2012-08-27 21:20:30 +01:00
Olivier Crête
264bcf7d6f rtph264pay: Make it actually work after cleanups 2012-08-08 19:49:05 -07:00
Mark Nauwelaerts
1547fdbe5a rtpmparobustdepay: set correct data_size for generated dummy frame
... which prevents getting stuck in a loop if such one is needed.
2012-08-06 14:58:21 +02:00
Mark Nauwelaerts
3e1832f5a4 rtpmparobustdepay: improve and fix debug statement
... so it really informs about next rather than past frame.
2012-08-06 14:58:21 +02:00
Mark Nauwelaerts
31a1cb0a11 rtpmparobustdepay: update available bytewriter space when repositioning
... and add some more assert to catch potential surprises early on.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680558
2012-08-06 14:58:21 +02:00
Mark Nauwelaerts
0bf9d8c6a6 rtpmparobustdepay: modify buffer data rather than buffer itself 2012-07-26 16:34:52 +02:00
Mark Nauwelaerts
c40807f6aa rtpmparobustdepay: avoid leaking bytewriter instance 2012-07-26 16:34:52 +02:00
Wim Taymans
0ed9e07c5d h264depay: small cleanups 2012-07-25 12:49:07 +02:00
Wim Taymans
4b92022120 rtp: always use buffer lists 2012-07-23 16:42:56 +02:00
Patricia Muscalu
3dd99f06f4 rtpmp4vpay: always enable buffer-lists 2012-07-23 16:17:37 +02:00
Patricia Muscalu
15cce2dd26 rtpjpegpay: always enable buffer-lists 2012-07-23 16:15:59 +02:00
Wim Taymans
51371d26ee update for RTP buffer api changes 2012-07-17 16:38:27 +02:00
Patricia Muscalu
d38ac43a27 rtph264pay: use buffer lists
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679994
2012-07-17 10:10:14 +02:00
Tim-Philipp Müller
c6224443a4 rtph264pay: avoid some relocations 2012-07-06 19:11:02 +01:00
Tim-Philipp Müller
3ef35ecdbc rtpmp4vpay: remove deprecated send-config property
Use config-interval instead.
2012-07-06 14:49:18 +01:00
Tim-Philipp Müller
cd1da84bcc rtph264depay: remove deprecated "byte-stream" and "access-unit" properties
These will be picked automatically based on downstream caps now, so
if you want the depayloader to output a specific format, make sure
the element downstream advertises that preference or use a capsfilter
after the depayloader to force it.
2012-07-06 14:46:22 +01:00
Tim-Philipp Müller
cffbf8cfc3 rtph264pay: remove deprecated and non-functional "profile-level-id" property
This is now optionally taken from downstream caps, so can be
specified via a capsfilter after the payloader.
2012-07-06 14:46:22 +01:00
Tim-Philipp Müller
48706beb70 rtph263ppay: accept any h263 input unless downstream forces specific requirements
rtph263ppay should accept any input compatible with its sink template
caps if it just outputs to e.g. udpsink or fakesink.

rtph263ppay ! rtph263pdepay should also work with any compatible input.
This would fail before with not-negotiated errors because the get_caps
function would see the encoding-name in the depayloader's template caps
and default to baseline H.263 because there's no profile/level information
in those caps, which is the right thing to do if downstream has filtercaps
from an SDP, but not if those fields are absent because they can be
anything like with the depayloader's template caps. Makes

  videotestsrc ! avenc_h263p ! rtph263ppay ! rtph263pdepay ! fakesink

work.
2012-07-06 11:57:38 +01:00
Wim Taymans
8eadb9c12c update for query api changes 2012-07-06 11:26:46 +02:00
Javier Jardón
c740490c26 rtp: remove some outdated comments
https://bugzilla.gnome.org/show_bug.cgi?id=679301
2012-07-03 08:58:26 +01:00
Wim Taymans
6d158775bb rtph264pay: cleanups
Use the caps properties for alignment and format.
Remove some old properties, we always want to use bufferlists when we can now.
2012-06-28 12:00:09 +02:00
Wim Taymans
429bda6923 h264pay: prefer AVC, it's easier to parse etc 2012-06-28 11:32:03 +02:00
Wim Taymans
540245894f theoradepay: fix buffer memory
The memory was added to the input buffer instead of the output buffer.
2012-06-14 10:43:56 +02:00
Thiago Santos
78ec03e32f Some printf variable format fixes
The osx compiler complains about those
2012-06-05 17:53:57 -03:00
Edward Hervey
923be8a85b rtpmp2tdepay: Only output integral mpeg-ts packets
From RFC 2250

2. Encapsulation of MPEG System and Transport Streams
...
   For MPEG2 Transport Streams the RTP payload will contain an integral
   number of MPEG transport packets.  To avoid end system
   inefficiencies, data from multiple small MTS packets (normally fixed
   in size at 188 bytes) are aggregated into a single RTP packet.  The
   number of transport packets contained is computed by dividing RTP
   payload length by the length of an MTS packet (188).
....

Since it needs to contain "an integral number of MPEG transport packets", a
simple fix is to check that's the case, and strip off any leftover data.

Fixes #676799

Conflicts:

	gst/rtp/gstrtpmp2tdepay.c
2012-05-26 12:04:54 +02:00
Luis de Bethencourt
c81fff0471 rtp: fix build issue in gstrtph264pay.c 2012-05-24 09:29:25 +01:00
Jonas Holmberg
7bf3a1bf95 rtph264pay: Add unrestricted caps
If there are no profile restrictions downstream, return caps with
profile=constrained-baseline in the first structure and append
unrestricted caps as the last structure.

Fixes bug #672019
2012-05-24 10:01:19 +02:00
Mark Nauwelaerts
182596b3ab rtpmp2tpay: respect mtu and packet boundaries
See #659915.
2012-05-18 12:53:44 +02:00
Youness Alaoui
7703a11073 rtpjpegpay: Allow U and V components to use different quant tables if they contain the same data
This allows some cameras (Logitech C920) that specify different quant
tables but both with the same data, to work.
Bug reported by Robert Krakora
2012-05-16 09:49:08 +02:00
idc-dragon
e0945d0a2d celtdepay: calculate size correctly
The summation was done wrong, causing the de-payloader to exit its loop too
early, before all frames are processed.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=674472
2012-04-25 10:29:56 +02:00
Sebastian Dröge
04b70571e5 video: Update for libgstvideo API changes 2012-04-19 12:20:59 +02:00
Edward Hervey
71fc25849e rtp: Use unchecked variant of GstByteWriter where applicable
The size was checked before
2012-04-12 15:50:16 +02:00
Tim-Philipp Müller
e09ae5736d Use new gst_element_class_set_static_metadata() 2012-04-10 00:51:41 +01:00
Sebastian Dröge
aa2cd462da gst: Update for GST_PLUGIN_DEFINE() API changes 2012-04-05 17:36:38 +02:00
Sebastian Dröge
5cdd49bf25 gst: Update versioning 2012-04-04 14:37:47 +02:00
Wim Taymans
3d61d12e03 update for buffer api change 2012-03-30 18:15:34 +02:00
Wim Taymans
69002aa24f update for buffer changes 2012-03-28 12:53:05 +02:00
Wim Taymans
e310ee8218 caps: improve caps handling
Avoid caps copy and leaks
2012-03-27 16:42:41 +02:00
Mark Nauwelaerts
e5ab3cc0a0 rtph264pay: ensure output caps are set when pushing output data
... even if some SPS/PPS has not passed by yet.
2012-03-26 18:38:34 +02:00
Mark Nauwelaerts
4bbc2a7106 rtpL16(de)pay: fix raw audio format in template caps 2012-03-26 18:38:34 +02:00
Olivier Crête
06f1c1817e rtph264depay: Make output in AVC stream format work even without complete sprop-parameter-set
This allows outputting streams in AVC format even if the SPS/PPS are sent inside
the RTP stream.

https://bugzilla.gnome.org/show_bug.cgi?id=654850

Ported from master
2012-03-22 16:18:37 -04:00
Wim Taymans
c44cd8f55b Merge branch 'master' into 0.11
unport gdkpixbuf
not merged: https://bugzilla.gnome.org/show_bug.cgi?id=654850

Conflicts:
	docs/plugins/Makefile.am
	docs/plugins/gst-plugins-good-plugins-docs.sgml
	docs/plugins/gst-plugins-good-plugins-sections.txt
	docs/plugins/gst-plugins-good-plugins.hierarchy
	docs/plugins/inspect/plugin-avi.xml
	docs/plugins/inspect/plugin-png.xml
	ext/flac/gstflacdec.c
	ext/flac/gstflacdec.h
	ext/libpng/gstpngdec.c
	ext/libpng/gstpngenc.c
	ext/speex/gstspeexdec.c
	gst/audioparsers/gstflacparse.c
	gst/flv/gstflvmux.c
	gst/rtp/gstrtpdvdepay.c
	gst/rtp/gstrtph264depay.c
2012-03-22 11:53:24 +01:00
Wim Taymans
ced47580b7 update for bufferpool changes 2012-03-15 22:11:17 +01:00
Wim Taymans
f3a770a20c update for allocation query changes 2012-03-15 20:37:56 +01:00
Olivier Crête
053f33adc8 rtph264depay: Make output in AVC stream format work even without complete sprop-parameter-set
This allows outputting streams in AVC format even if the SPS/PPS are sent inside
the RTP stream.

https://bugzilla.gnome.org/show_bug.cgi?id=654850
2012-03-15 14:20:22 -04:00
Wim Taymans
751fcf035b take padding into account 2012-03-14 19:56:56 +01:00
Wim Taymans
734f11e4d3 mp4vpay: we can also handle x-divx 2012-03-14 11:26:35 +01:00
Wim Taymans
fba47d17e8 mp4vdepay: fix buffer handling
Don't always output the payload subbuffer, use a separate variable to
make things clearer and without the error.
2012-03-13 21:31:48 +01:00
Wim Taymans
745210e792 h264depay: unmap on empty packet 2012-03-13 19:26:23 +01:00
Wim Taymans
d65de434f5 rtph264pay: do DTS and PTS correctly 2012-03-13 18:07:18 +01:00
Wim Taymans
e4fed38f49 rtp: fix unmap calls 2012-03-13 17:27:32 +01:00
Wim Taymans
a32d944a38 fix for caps api changes 2012-03-11 19:06:37 +01:00
Sebastian Dröge
78079635a6 dvdepay: Fix 'comparison of unsigned expression >= 0 is always true' compiler warning
This was an actual bug as it could've caused reading from
invalid memory areas when the input is broken.
2012-03-06 14:16:21 +01:00
Wim Taymans
ca9532ccc5 update for new memory api 2012-02-22 02:10:33 +01:00
Olivier Crête
18899cf94d rtph264pay: Force baseline is profile-level-id is unspecified 2012-02-21 10:51:43 +01:00
Olivier Crête
1fe69911a4 rtph264pay: Force baseline is profile-level-id is unspecified 2012-02-20 14:30:55 -05:00
Wim Taymans
225e98d623 Merge branch 'master' into 0.11
Conflicts:
	ext/flac/gstflacenc.c
	ext/jack/gstjackaudioclient.c
	ext/jack/gstjackaudiosink.c
	ext/jack/gstjackaudiosrc.c
	ext/pulse/plugin.c
	ext/shout2/gstshout2.c
	gst/matroska/matroska-mux.c
	gst/rtp/gstrtph264pay.c
2012-02-10 16:23:14 +01:00
Tim-Philipp Müller
5b25f3737b rtph264pay: add stream-format and alignment to h264 sink caps
We're happy to accept both byte-stream and avc, advertise
that on the sink caps and fix up _get_caps() function to
not just return "video/x-h264".

https://bugzilla.gnome.org/show_bug.cgi?id=606662
2012-02-10 14:08:55 +00:00
Tim-Philipp Müller
6872b40873 rtph264depay: add stream-format and alignment fields to src template caps
Because we can. And so we get a warning if we try to output avc with
nal alignment or somesuch.

https://bugzilla.gnome.org/show_bug.cgi?id=606662
2012-02-10 14:08:55 +00:00
Vincent Penquerc'h
d651baf05a rtpmp2tpay: do not try to flush a packet when no data is available
https://bugzilla.gnome.org/show_bug.cgi?id=668874
2012-01-31 13:12:47 +00:00
Pascal Buhler
c16fed2ad9 rtph264depay: Exclude NALu size from payload length on truncated packets.
https://bugzilla.gnome.org/show_bug.cgi?id=667846
2012-01-30 15:49:07 +00:00
Sebastian Dröge
0b517ce9fb Merge branch '0.11' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-good into 0.11 2012-01-25 12:49:34 +01:00
Sebastian Dröge
10554b271f Merge branch 'master' into 0.11
Conflicts:
	ext/flac/gstflacdec.c
	ext/jpeg/gstjpegenc.c
	ext/pulse/pulsesink.c
	sys/v4l2/gstv4l2src.c
2012-01-25 12:49:11 +01:00
Wim Taymans
b4630dd3e0 more memory API porting 2012-01-25 12:30:29 +01:00
Wim Taymans
583d39dd8d update for new memory API 2012-01-25 12:30:28 +01:00
Mark Nauwelaerts
a3ea25bc88 rtpmp4adepay: prevent out-of-bound array access 2012-01-20 17:10:48 +01:00
Mark Nauwelaerts
ed94e01231 rtptheoradepay: remove dead code 2012-01-20 17:10:40 +01:00
Sebastian Dröge
59e08fa503 configure: Remove socket/winsock specific checks
Not necessary anymore.
2012-01-17 16:53:31 +01:00
Vincent Penquerc'h
2a7a38ca07 rtph263ppay: fix caps leak 2012-01-16 15:42:46 +00:00
Sebastian Dröge
4885f34458 rtp: Update for the new audio caps 2012-01-05 10:30:34 +01:00
Tim-Philipp Müller
668e15598b Merge remote-tracking branch 'origin/master' into 0.11
Conflicts:
	sys/v4l2/gstv4l2object.c
2011-12-08 01:28:26 +00:00
Wim Taymans
b1d771cf8c h263pay: fix invalid return value 2011-12-06 14:23:30 +01:00
Edward Hervey
04520cbe9a rtp: Initialize GstRTPBuffer before usage 2011-12-05 18:39:59 +01:00
Sebastian Rasmussen
c090201ca5 rtpjpegpay: Ceil jpeg dimensions, instead of floor
A JPEG image inside an RTP stream has a preceeding RFC2435 header that
conveys width/height. The dimensions in this header are limited to be
multiples of 8. Since JPEG uses an MCU of 8x8 pixels any image must
already indirectly have image data dimensions that are rounded up in
order to contain enough data to render the image. Therefore this fix
safely rounds the image dimensions in the RFC2435 header up to the
closest multiple of 8.
2011-12-05 10:48:54 +01:00
Vincent Penquerc'h
c0e101e93f various: fix pad template leaks
https://bugzilla.gnome.org/show_bug.cgi?id=662664
2011-11-28 13:30:27 +00:00
Matej Knopp
1e5dd9e315 Fix printf format compiler warnings on OS X / 64bit
https://bugzilla.gnome.org/show_bug.cgi?id=662615
2011-11-22 01:28:22 +00:00
Wim Taymans
105650127e add parent to pad functions 2011-11-17 15:02:55 +01:00
Wim Taymans
797523efbd _peer_get_caps() -> _peer_query_caps() 2011-11-15 18:04:44 +01:00
Wim Taymans
75dc9634eb change getcaps to query
Chain up event function in payloaders.
2011-11-15 18:04:44 +01:00
Wim Taymans
af1eec2ece rtp: fix for rtp header changes 2011-11-11 19:21:50 +01:00
Wim Taymans
e84b8dbe94 update for base class rename 2011-11-11 12:32:41 +01:00
Wim Taymans
249d0083cc update for base class rename 2011-11-11 12:25:01 +01:00
Wim Taymans
7e12b58e37 update for adapter api changes 2011-11-10 18:32:58 +01:00
Wim Taymans
fbaf216d25 update for changed base classes 2011-11-10 17:23:47 +01:00
Wim Taymans
85e73e0818 h263ppay: report to 0.11 2011-11-09 12:25:01 +01:00
Wim Taymans
95f3987332 Merge branch 'master' into 0.11
Conflicts:
	ext/flac/gstflacdec.c
	gst/audioparsers/gstflacparse.c
	gst/isomp4/qtdemux.c
2011-11-09 12:18:01 +01:00
Olivier Crête
e15c293f13 rtph263ppay: Return the sink pad template as sink caps, not the src's
https://bugzilla.gnome.org/show_bug.cgi?id=577784
2011-11-08 15:53:39 +01:00
Olivier Crête
4b28d9d44e rtph263ppay: Also implement size/framerate restrictions in getcaps
https://bugzilla.gnome.org/show_bug.cgi?id=577784
2011-11-08 15:53:18 +01:00
Olivier Crête
ff31090671 rtph263ppay: Implement getcaps following RFC 4629, picks the right annexes
https://bugzilla.gnome.org/show_bug.cgi?id=577784
2011-11-08 15:52:57 +01:00
Tim-Philipp Müller
d65490dfad rtp: use GLib's G_BIG_ENDIAN define instead of BIG_ENDIAN
Fixes compiler warning on mingw32
2011-11-03 23:28:31 +00:00
Wim Taymans
b1ef7e8a86 update for meta api change 2011-11-02 09:06:37 +01:00
Wim Taymans
9a8a8e72c8 structure: fix for api update 2011-11-02 09:06:37 +01:00
Wim Taymans
9c14280b1d make some more things compile again 2011-10-27 19:00:52 +02:00
Wim Taymans
fc4684f4c6 fix compilation 2011-10-27 16:03:17 +02:00
Marc Leeman
98075ad70d set colour masks for video/x-raw-rgb in rtpvrawdepay 2011-10-14 09:32:47 +02:00
Wim Taymans
a5cc912140 Merge branch 'master' into 0.11
Conflicts:
	ext/jpeg/gstjpegdec.c
	gst/rtp/gstrtpvrawpay.c
2011-10-13 08:58:06 +02:00
Edward Hervey
1b56d40170 rtpvrawpay: Only use 24 LSB for depth=24 RGB caps
... and indent the masks for clarity
2011-10-12 11:26:50 +02:00
Sjoerd Simons
bf65acf11f gstrtpg722pay: Compensate for clockrate vs. samplerate difference
The RTP clock-rate used for G722 is 8000, even though the samplerate is
16000. Compensate for this by pretending G722 has 8 bits per sample
instead of the 4 bits as if it were a codec that ran at half the speed,
but with twice the number of bits. Fixes #661376
2011-10-10 21:50:28 +01:00
Wim Taymans
87fbd1e784 Merge branch 'master' into 0.11
Conflicts:
	common
	ext/pulse/pulsesink.c
	ext/soup/gstsouphttpclientsink.c
	gst/audioparsers/gstaacparse.c
	gst/audioparsers/gstac3parse.c
	gst/rtp/gstrtph264depay.c
	gst/rtpmanager/gstrtpjitterbuffer.c
	gst/rtpmanager/rtpjitterbuffer.c
	gst/rtsp/gstrtspsrc.c
	sys/ximage/gstximagesrc.c
2011-09-28 12:44:59 +02:00
Mark Nauwelaerts
fd757890eb rtph264depay: improve downstream flow return feedback to upstream
... although basertpdepay does not really make it easy/possible to do so
all the way.
2011-09-20 14:14:39 +02:00
Wim Taymans
83ea243000 Merge branch 'master' into 0.11
Conflicts:
	common
2011-09-06 16:37:03 +02:00
Wim Taymans
33f18b8ea4 Merge branch 'master' into 0.11
Conflicts:
	gst/audioparsers/gstamrparse.c
	gst/isomp4/qtdemux.c
2011-09-06 16:06:25 +02:00
Mark Nauwelaerts
06f8e356a6 rtpmp4gdepay: improve bogus interleaved index compensating
Patch by <gudake@gmail.com>

Fixes #654585.
2011-09-06 13:20:23 +02:00
Olivier Crête
d4778dbe43 rtph263ppay: Set H263-2000 if thats what the other side wants
The static caps states this element supports H263-2000, but setcaps never
sets it, so it was lie.

See https://bugzilla.gnome.org/show_bug.cgi?id=577784
2011-09-05 12:58:55 +02:00
Wim Taymans
24df106272 mp2t: fix encoding name according to RFC3551 2011-08-31 18:45:15 +02:00
Wim Taymans
18065ac823 port to new video flags 2011-08-25 16:41:23 +02:00
Wim Taymans
60f0e44bf6 video: port to new colorimetry info 2011-08-23 19:09:31 +02:00
Wim Taymans
9d6371405e fourcc: remove fourcc from caps 2011-08-22 12:24:15 +02:00
Wim Taymans
77ad0a1363 port more elements to new audio caps and API 2011-08-19 14:01:45 +02:00
Wim Taymans
ee2aa25e04 port to new API 2011-08-03 18:37:27 +02:00
Wim Taymans
4121021bb2 Merge branch 'master' into 0.11
Conflicts:
	ext/pulse/pulsesink.c
	ext/pulse/pulsesrc.c
	gst/audioparsers/gstac3parse.c
	gst/rtp/gstrtph264depay.c
	gst/rtp/gstrtph264pay.c
	gst/rtpmanager/gstrtpssrcdemux.c
2011-08-03 18:25:30 +02:00
Robert Krakora
f7893b8721 rtpjpegpay: Add support for H.264 payload in MJPEG container
See http://www.quickcamteam.net/uvc-h264/USB_Video_Payload_H.264_0.87.pdf

Fixes bug #655530.
2011-08-03 10:09:42 +02:00
Wim Taymans
5771056ed5 rtpvorbispay: fix porting error 2011-08-02 11:51:45 +02:00
Wim Taymans
49af68ebf4 -good: fix for bufferpool API change 2011-07-29 17:27:07 +02:00
Sjoerd Simons
4c73439ee3 rtph264depay: Cope with FU-A E bit not being set
Some h264 payloaders are unfortunately buggy and don't correctly set the
E bit in FU-A NAL when they have ended. Work around this by assuming
such a fragmentation unit has ended when there was no packet loss and a
new NAL is started
2011-07-27 18:18:13 +01:00
Wim Taymans
3e089bd7a9 rtp: fix compilation 2011-07-26 17:45:01 +02:00
Olivier Crête
2591a882ae rtph264depay: Complete merged AU on marker bit
The marker bit on a RTP packet means the AU has been completed, so push it out
immediately to reduce the latency.

https://bugzilla.gnome.org/show_bug.cgi?id=654850
2011-07-21 17:11:08 +02:00
Olivier Crête
118a7cc36a rtph264pay: Only set the marker bit on the last NALU of a multi-NALU access unit
An access unit could contain multiple NAL units, in that case, only the last
RTP packet of the last NALU should have its marker bit set.

https://bugzilla.gnome.org/show_bug.cgi?id=654850
2011-07-21 17:11:06 +02:00
Mark Nauwelaerts
471904032d rtph264depay: reset upon FLUSH_STOP
... which is particularly needed when merging NAL units, where not resetting
would lead to output of an older (pre-flush) AU (with unintended timestamp).
2011-07-18 14:32:26 +02:00
Wim Taymans
9c087d7d85 Merge branch 'master' into 0.11 2011-07-15 17:06:39 +02:00
Olivier Crête
87c7f303b0 rtppcmApay/depay: Static clock rates on static payloads, dynamic on dynamic
Partially reverts 397dc60b
2011-07-14 20:13:01 -04:00
Olivier Crête
57a832cbb1 rtph264pay: Implement getcaps
Convert profile-level-id from RTP caps into video/x-h264 style caps (with profile and level)
2011-07-13 14:10:35 -04:00
Mark Nauwelaerts
eb82a50bd1 rtp: port remaining to 0.11 2011-07-10 21:50:19 +02:00
Wim Taymans
cc65bff7c1 Merge branch 'master' into 0.11
Conflicts:
	configure.ac
	docs/plugins/inspect/plugin-esdsink.xml
	docs/plugins/inspect/plugin-gconfelements.xml
2011-06-21 18:24:41 +02:00
Mark Nauwelaerts
3daf1ecc21 rtpmp4adepay: fix output buffer timestamps in case of multiple frames 2011-06-21 15:15:33 +02:00
Wim Taymans
3c889415a3 rtp: port some more (de)payloader 2011-06-13 17:14:00 +02:00
Wim Taymans
9a54175e9f rtp: port to 0.11 2011-06-13 16:33:46 +02:00
Wim Taymans
b0fbb1725f rtp: fix for API changes in the base classes 2011-06-13 13:25:49 +02:00
Wim Taymans
0b1bdcf7cb Merge branch 'master' into 0.11
Conflicts:
	sys/ximage/ximageutil.c
2011-06-02 18:51:29 +02:00
Marc Leeman
ff1c05d876 rtpmp4vpay: Deprecated send-config property and replace by config-interval
Fixes bug #622412.
2011-05-26 12:22:52 +02:00
Wim Taymans
d89790d545 Merge branch 'master' into 0.11
Conflicts:
	gst/avi/gstavidemux.c
	gst/rtp/gstrtpac3depay.c
	gst/rtp/gstrtpg726depay.c
	gst/rtp/gstrtpmpvdepay.c
	gst/videofilter/gstgamma.c
2011-05-24 17:34:19 +02:00
Mark Nauwelaerts
397dc60b71 pcmudepay: allow variable sample rate 2011-05-24 13:13:55 +02:00
Mark Nauwelaerts
f335fee99e pcmadepay: allow variable sample rate 2011-05-24 13:13:52 +02:00
Stefan Kost
d122ea0122 rtp: fix static array overruns in a nicer way
Use G_N_ELEMENTS instead of hard-coding the array size.
2011-05-20 10:34:47 +03:00
Stefan Kost
5792d3b9c0 rtp: fix static array overruns
Yes array[10] has elements from 0...9.
2011-05-20 00:53:44 +03:00
Jose Antonio Santos Cadenas
9d32243671 rtp: Fix segmentation fault processing payload buffers
This commit checks if the value returned by
gst_rtp_buffer_get_payload_buffer and
gst_rtp_buffer_get_payload_subbuffer is NULL before using it.
2011-05-18 15:25:24 +02:00
Wim Taymans
31ffc671f2 rtpgstpay: fix buffer leak 2011-04-26 16:04:07 +01:00
Wim Taymans
eb84592cad rtpgstpay: fix buffer leak 2011-04-26 15:58:12 +02:00
Wim Taymans
9a96783abb rtp: port some more elements 2011-04-25 18:14:45 +02:00
Wim Taymans
bf9b4f8362 rtp: port more to 0.11 2011-04-25 17:27:40 +02:00
Wim Taymans
60db07b4bb rtp: port some more (de)payloaders 2011-04-25 13:16:58 +02:00
Wim Taymans
4aa6ca5578 port more plugins to 0.11 2011-04-18 10:54:43 +02:00
Wim Taymans
7555d0949f Merge branch 'master' into 0.11
Conflicts:
	android/apetag.mk
	android/avi.mk
	android/flv.mk
	android/icydemux.mk
	android/id3demux.mk
	android/qtdemux.mk
	android/rtp.mk
	android/rtpmanager.mk
	android/rtsp.mk
	android/soup.mk
	android/udp.mk
	android/wavenc.mk
	android/wavparse.mk
	configure.ac
2011-04-18 10:23:45 +02:00
Tim-Philipp Müller
f325935314 pulse, speexenc, rtpgsmpay: don't use g_assert() for error handling
Don't use g_assert() for error handling, even if they're highly unlikely.
Either we *know* that something can't happen, in which case we
should just not handle it, or we think something can happen, but it is
very very unlikely that it will ever happen, in which case we should
handle it like any other error instead of asserting.

g_assert() is best left for conditions we have control of, like checking
internal consistency of our code, not checking return values of external
code.

Fixes a bunch of warnings when compiling with -DG_DISABLE_ASSERT:
gstrtpgsmpay.c: In function 'gst_rtp_gsm_pay_handle_buffer':
gstrtpgsmpay.c:130:17: warning: variable 'rtpgsmpay' set but not used
gstspeexenc.c: In function 'gst_speex_enc_encode':
gstspeexenc.c:904:19: warning: variable 'written' set but not used
pulsesink.c: In function 'gst_pulsesink_change_state':
pulsesink.c:2725:9: warning: variable 'res' set but not used
pulsesrc.c: In function 'gst_pulsesrc_change_state':
pulsesrc.c:1253:7: warning: variable 'e' set but not used
2011-04-16 18:15:43 +01:00
Robert Swain
5b18c652fb rtp, rtpmanager: Address unused but set variables
GCC 4.6.x spits warnings about variables that are unused but set. Such
variables have been removed where trivial but with comments left behind
for informational purposes in some cases.

gst_rtp_session_chain_recv_rtcp () was changed in commit 490113d4
to always return GST_FLOW_OK instead of the return value of
rtp_session_process_rtcp (), so we'll keep it that way.
2011-04-16 12:49:16 +01:00