Commit graph

189 commits

Author SHA1 Message Date
Sebastian Dröge
df9b8b57b3 introspection: Build pkgconfig before all libraries and set PKG_CONFIG_PATH
This way g-ir-scanner can find the gstreamer-*-0.10 pkg-config files.
2009-09-13 11:19:50 +02:00
Wim Taymans
e2e7ae0129 basertppay: don't print RTP timestamps as clocktime
Don't try to print the RTP timestamp as a GstClockTime, it's just a guint32.

Fixes #594757
2009-09-10 18:21:08 +02:00
Havard Graff
f710bec408 basertpdepayload: fix event forwarding 2009-09-08 15:10:59 +02:00
Havard Graff
f0f72088bc rtcpbuffer: add missing break in handling of GST_RTCP_TYPE_PSFB
Fixes #594258
2009-09-08 13:03:21 +02:00
Sebastian Dröge
7e90e0846c introspection: Strip Gst prefix from all types/functions 2009-09-05 12:31:47 +02:00
Sebastian Dröge
7794caf9f8 introspection: Fix build if gir-repository is not installed 2009-09-05 11:49:41 +02:00
Sebastian Dröge
8001b380b1 rtp: Add gobject-introspection support 2009-09-05 11:25:42 +02:00
Wim Taymans
7a7663476f audiortppay: add some debugging 2009-09-03 18:53:19 +02:00
Wim Taymans
c1db9ebb20 audiortppay: handle gaps
Add various conversion functions between time<->bytes<->rtptime that will be
used later on.
Refactor the min/max packet length code so that it can be used for both
sample/frame based payloaders. Cache the returned values.
code cleanups.
When we discover a DISCONT buffer, make the outgoing RTP timestamps have the
same gap as the GStreamer timestamps gap.
2009-09-03 17:59:00 +02:00
Wim Taymans
3a3c6f309c audiortppay: fix frame duration calculations
Fix the calculation of the frame duration and rtp timestamps.
Add some debugging
2009-09-03 17:59:00 +02:00
Wim Taymans
bfc19462bb rtppay: add some debugging 2009-09-03 17:59:00 +02:00
Wim Taymans
bb91a7b47c audiortppay: use offsets for RTP timestamps
Have a custom sample/frame function to generate an offset that the base class
will use for generating RTP timestamps. This results in perfect RTP timestamps
on the output buffers.
Refactor setting metadata on output buffers.
Add some more functionality to _flush().
Handle DISCONT on the input buffers and set the marker bit and DISCONT flag on
the next outgoing buffer.
Flush the pending data on EOS.
2009-09-03 17:58:59 +02:00
Wim Taymans
c1ae0a2003 audiortppay: move function around 2009-09-03 17:58:59 +02:00
Wim Taymans
5808041f44 audiortppay: fix sample duration calculation 2009-09-03 17:58:59 +02:00
Wim Taymans
299ab7be0e audiortppay: more refactoring
Unify the sample/frame buffer handling code by making the functions plugable.
2009-09-03 17:58:59 +02:00
Wim Taymans
fb5037f727 audiortppayload: refactor some more
Refactor getting the packet min/max size and alignment code.
Refactor converting bytes to time.
change some variable to something shorter.
2009-09-03 17:58:59 +02:00
Wim Taymans
1c6b71af03 audiortppayload: refactor and cleanup
Always use the adapter when we need to fragment the incomming buffer. Use more
modern adapter functions to avoid malloc and memcpy. The overall result is that
the code looks cleaner while it should be equally fast and in some case avoid a
memcpy and malloc.
Use the adapter timestamping functions for more precise timestamps in case of
weird disconts.
Cache some values instead of recalculating them.
Add gst_base_rtp_audio_payload_flush() to flush a certain amount of bytes from
the internal adapter.

API: GstBaseRTPAudioPayload::gst_base_rtp_audio_payload_flush()
2009-09-03 17:58:59 +02:00
Wim Taymans
50b9640d01 basertppay: add property to disable perfect RTP time
Add a property to disable the generation of perfect RTP timestamps. By default
it is active.

API: GstBaseRTPPayload::perfect-rtptime
2009-09-03 11:29:23 +02:00
Wim Taymans
3a4edea56d basertppay: allow subclasses to influence RTP time
Allow subclasses to use the OFFSET field on RTP buffers to influence the way in
which RTP timestamps are generated. Usually timestamps are created from the
GStreamer timestamps on the buffer, which could result in imperfect RTP
timestamps.
2009-09-03 11:15:20 +02:00
Wim Taymans
5a479669d4 basertppay: add macro to cast 2009-09-03 11:15:20 +02:00
Wim Taymans
bc3c8a1564 audiopayload: code cleanups 2009-09-03 11:15:20 +02:00
Wim Taymans
3c29efa692 audiortppayload: don't check adapter
the adapter is never NULL so we don't need to check it.
Use _scale functions to avoid overflows.
2009-09-03 11:15:20 +02:00
Jonas Holmberg
ec91d508af basertppayload: Make instance init faster by not reading /dev/urandom 3 times
... which is the default seed when creating a new GRand. Because
GLib in older versions used buffered IO this would take a lot of time.

Instead use the global GRand for getting random numbers and keep the
three instance GRand for backward compatibility with a simple seed.

Fixes bug #593284.
2009-09-01 10:39:52 +02:00
Wim Taymans
e40b262ab7 basertppayload: whitespace fixes. 2009-08-28 14:09:02 +02:00
Edward Hervey
9329b8be72 gst-libs: Remove dead assignments and resulting unused variables. 2009-08-08 15:54:57 +02:00
Wim Taymans
85af9b82e8 basertppayload: add support for bufferlists
Based on patch from Ognyan Tonchev.

See #585559
2009-06-19 15:52:34 +02:00
Wim Taymans
f5c8055edf rtpbuffer: use new convenience functions
New core convenience functions makes the list getters and setters trivial.
Maybe even too trivial...
2009-06-19 15:33:04 +02:00
Wim Taymans
457d39075c rtp: cleanups, add _list_get_seq() too
Clean up the docs a little.
Add missing _list_get_seq method.
Add new symbols to the docs
2009-06-18 19:04:52 +02:00
Wim Taymans
e2ccc1ee39 rtp: cleanups
Add Since tags to docs
Move some code around
Add win32 symbols
2009-06-18 18:51:04 +02:00
Wim Taymans
66c388a0e0 rtp: add bufferlist support 2009-06-18 18:51:04 +02:00
Wim Taymans
f385081c92 rtp: pass data to macros instead of GstBuffer 2009-06-18 18:50:35 +02:00
Wim Taymans
8bbf2e8a32 docs: fix typo 2009-06-11 12:39:19 +02:00
Wim Taymans
0a09632396 rtpdepay: add some more comments 2009-05-12 10:39:49 +02:00
Olivier Crete
d927114ef8 RTCP: don't fail when retrieving invalid PT
We can't meaningfully assert on valid packet types so just return the type as it
is. Update the comments to reflect this.

Fixes #579192.
2009-04-17 10:53:10 +02:00
Johann Prieur
86edcadc43 RTCP: add beginnings of Feedback messages
Add the beginnings of parsing and constructing Feedback messages.
Fixes #577610.
2009-04-14 16:45:20 +02:00
Stefan Kost
251e4d160a docs: don't put random stuff in tags.
Tags like Since: or Returns: can only be followed by more tags. gtk-doc has no
tag to append text again to the documentation body.
2009-02-26 10:09:59 +02:00
Wim Taymans
9ce042e2a7 Avoid overflows in the padding checks by doing the check slightly
differently.
Add a unit test to check for correct behaviour.
2009-01-21 13:09:29 +01:00
Olivier Crete
3c9df39c15 gst-libs/gst/rtp/gstrtcpbuffer.*: Implement gst_rtcp_packet_remove(). Fixes #563174.
Original commit message from CVS:
Patch by: Olivier Crete  <tester at tester ca>
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_remove):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Implement gst_rtcp_packet_remove(). Fixes #563174.
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add unit test for some RTCP functions.
2008-12-08 12:08:32 +00:00
Wim Taymans
b2004e3d05 gst-libs/gst/rtp/gstrtcpbuffer.c: Fix typo in the docs.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
Fix typo in the docs.
2008-11-25 15:33:30 +00:00
Wim Taymans
9c32e1f152 gst-libs/gst/rtp/gstrtpbuffer.c: Avoid expensive type checks we already did as part of the _validate() function that ...
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
(gst_rtp_buffer_set_packet_len), (gst_rtp_buffer_get_packet_len),
(gst_rtp_buffer_get_header_len), (gst_rtp_buffer_get_version),
(gst_rtp_buffer_set_version), (gst_rtp_buffer_get_padding),
(gst_rtp_buffer_set_padding), (gst_rtp_buffer_pad_to),
(gst_rtp_buffer_get_extension), (gst_rtp_buffer_set_extension),
(gst_rtp_buffer_get_extension_data),
(gst_rtp_buffer_set_extension_data), (gst_rtp_buffer_get_ssrc),
(gst_rtp_buffer_set_ssrc), (gst_rtp_buffer_get_csrc_count),
(gst_rtp_buffer_get_csrc), (gst_rtp_buffer_set_csrc),
(gst_rtp_buffer_get_marker), (gst_rtp_buffer_set_marker),
(gst_rtp_buffer_get_payload_type),
(gst_rtp_buffer_set_payload_type), (gst_rtp_buffer_get_seq),
(gst_rtp_buffer_set_seq), (gst_rtp_buffer_get_timestamp),
(gst_rtp_buffer_set_timestamp),
(gst_rtp_buffer_get_payload_subbuffer),
(gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload):
Avoid expensive type checks we already did as part of the
_validate() function that should be called first.
2008-11-13 15:37:40 +00:00
Wim Taymans
c98d4a5031 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix some cases where a newsegment event was not sent.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c: (create_segment_event),
(gst_base_rtp_depayload_push_full),
(gst_base_rtp_depayload_set_gst_timestamp):
Fix some cases where a newsegment event was not sent.
2008-11-11 16:40:50 +00:00
Wim Taymans
4ae82906ab gst-libs/gst/rtp/gstbasertpdepayload.*: Add some more G_LIKELY
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_change_state):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Add some more G_LIKELY
Fail when the setcaps function was not called.
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_set_outcaps):
Propagate return value of setcaps.
2008-10-13 09:16:59 +00:00
Wim Taymans
a2eb053641 gst-libs/gst/rtp/gstbasertppayload.c: Configure the next seqnum and timestamp in the state change so that they can be...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_change_state):
Configure the next seqnum and timestamp in the state change so that they
can be queried soon after.
2008-10-06 16:31:27 +00:00
Wim Taymans
b86ef2dcf2 gst-libs/gst/rtp/gstbasertpdepayload.c: Improve debugging of the rtptime.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain):
Improve debugging of the rtptime.
2008-10-06 16:29:33 +00:00
Wim Taymans
11309247f3 gst-libs/gst/rtp/gstbasertppayload.c: Simply converting the running time into an RTP timestamp by scaling it based on...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push),
(gst_basertppayload_change_state):
Simply converting the running time into an RTP timestamp by scaling it
based on the clock-rate is good enough for making an RTP timestamp. This
has the added benefit that we can later on expose a property with the
RTP timestamp of running time 0, as is needed for RTSP servers to
generate the response of the PLAY request.
2008-05-30 15:29:20 +00:00
Wim Taymans
79a725148d gst-libs/gst/rtp/gstbasertpdepayload.c: Check sequence numbers, mark input buffers with a discont flag for the subcla...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_handle_sink_event),
(gst_base_rtp_depayload_push_full),
(gst_base_rtp_depayload_change_state):
Check sequence numbers, mark input buffers with a discont flag for the
subclass when we detected a gap, drop duplicate buffers. We do this
because one can use the element without a jitterbuffer in front and we
don't want to feed the subclasses invalid or reordered data.
Do an error when the subclass did not provide a process function instead
of crashing.
Some other small cleanups.
2008-05-23 14:14:28 +00:00
Wim Taymans
86ab51207b gst-libs/gst/rtp/gstbasertpdepayload.c: Validate the RTP packet before further processing it. It's just too dangerous...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain):
Validate the RTP packet before further processing it. It's just too
dangerous to accept random packets and people are not forced to use a
jitterbuffer or session manager to filter out the bad packets.
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_set_extension_data),
(gst_rtp_buffer_get_payload_subbuffer):
Small cleanups.
When setting extension data in a buffer that is too small, we fail and
we should not set the extension bit.
Change GST_WARNINGS into g_warning because they really are
programming errors.
* tests/check/libs/rtp.c: (GST_START_TEST):
Catch the g_warnings now in the unit tests and that fact that failing to
set extension data left the extension bit untouched.
2008-05-14 20:28:02 +00:00
Bernard B
d06df554a9 gst-libs/gst/rtp/gstrtpbuffer.c: Fix seqnum compare function for bordercase values and fix the docs again. Fixes #533...
Original commit message from CVS:
Patch by: Bernard B <b-gnome at largestprime dot net>
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum):
Fix seqnum compare function for bordercase values and fix the docs
again. Fixes #533075.
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add a testcase for seqnum compare function.
2008-05-14 13:43:12 +00:00
Wim Taymans
c6389eec57 gst-libs/gst/rtp/gstbasertppayload.c: Rename the setcaps/getcaps function internally to make it clear that they are c...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
(gst_basertppayload_sink_setcaps),
(gst_basertppayload_sink_getcaps):
Rename the setcaps/getcaps function internally to make it clear that
they are called for the sink pad.
2008-05-02 12:13:08 +00:00
Wim Taymans
f0f6476aff gst-libs/gst/rtp/gstbasertpdepayload.*: Catch packet-lost events from the jitterbuffer and convert them into a vmetho...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init),
(gst_base_rtp_depayload_handle_sink_event), (create_segment_event),
(gst_base_rtp_depayload_packet_lost),
(gst_base_rtp_depayload_set_gst_timestamp):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Catch packet-lost events from the jitterbuffer and convert them into a
vmethod call (lost-packet) so that depayloaders can do something smart.
Also add a default packet-lost function that sends out a segment update
to the decoders.
2008-05-02 12:11:07 +00:00