Commit graph

8163 commits

Author SHA1 Message Date
Sebastian Dröge
c60038f188 rtpsource: Queue bad packets instead of dropping them
So we can send them out once we found the next, consecutive sequence number in
case one is following.
2015-05-18 18:43:16 +03:00
Sebastian Dröge
9f18a271f3 rtpsource: Use g_queue_foreach() to unref all buffers in queues 2015-05-18 18:43:16 +03:00
Sebastian Dröge
54e924332e rtpsource: Refactor seqnum comparison code a bit 2015-05-18 18:43:16 +03:00
Sebastian Dröge
1974b24ef4 rtpsource: Allow sequence number wraparound during probation 2015-05-18 18:43:16 +03:00
Sebastian Dröge
3386de7a8a rtpsource: Make sequence number comparison code more readable
... by using gst_rtp_buffer_compare_seqnum() and signed integers
instead of implictly using effects of integer over/underflows.
2015-05-18 18:43:16 +03:00
Sebastian Dröge
ca110fb0b8 rtpjitterbuffer: When detecting a huge seqnum gap, wait for 5 consecutive packets before resetting everything
It might just be a late retransmission or spurious packet from elsewhere, but
resetting everything would mean that we will cause a noticeable hickup. Let's
get some confidence first that the sequence numbers changed for whatever
reason.

https://bugzilla.gnome.org/show_bug.cgi?id=747922
2015-05-18 18:43:15 +03:00
Nicolas Dufresne
1797c8f8b1 doc: Workaround gtkdoc issue
With gtkdoc 1.22, the XML generator fails when a itemizedlist is
followed by a refsect2. Workaround the issue by wrapping the
refsect2 into para.
2015-05-16 23:37:06 -04:00
Stefan Sauer
426eb3e300 qtdemux: avoid wrong warnings on unknown node types
Add 'name' and 'mean' fourccs, as we handle them. Right now each use would
trigger a warning.
2015-05-15 14:56:07 +02:00
Nicola Murino
fefeda5e6c rtpg726depay: add block_align to output caps
It is needed to correctly negotiate caps with matroskamux
and most other muxers.

https://bugzilla.gnome.org/show_bug.cgi?id=749129
2015-05-13 12:39:07 +01:00
Sebastian Dröge
e11a537b65 audiofxbasefirfilter: Fix time-domain convolution with >1 channels
input_samples is the number of frames, but we used it as the number of
samples.

https://bugzilla.gnome.org/show_bug.cgi?id=747204
2015-05-12 13:41:58 +03:00
Tim-Philipp Müller
2e412a447a docs: update example pipelines in element docs
Mostly gst-launch -> gst-launch-1.0
Use autovideosink/autoaudiosink more often.
Sprinkle some converters here and there.
2015-05-10 11:05:00 +01:00
Tim-Philipp Müller
3755409409 splitmuxsrc: minor error message clean-up
Don't put filename in error message shown to user.
2015-05-10 10:53:13 +01:00
Guillaume Desmottes
2bd3685d04 flacparse: fix buffer leak when stored to seektable
Fix a leak with the
validate.file.playback.change_state_intensive.samples_multimedia_cx_flac_Yesterday_flac
scenario.

https://bugzilla.gnome.org/show_bug.cgi?id=749072
2015-05-08 11:11:40 +01:00
Paul Hyunil
3792e9ca9b qtdemux: fix example pipeline in docs
The gst-launch script for example launch line to test qtdemux is
missing a queue before the decodebins, otherwise the gst-launch-1.0
command won't work.

https://bugzilla.gnome.org/show_bug.cgi?id=749054
2015-05-08 11:06:31 +01:00
Sebastian Dröge
27729a2960 Revert "rtpsession: Also report internal sources in on-new-ssrc and on-ssrc-active"
This reverts commit d22ec49632.

Application code might expect that it only gets external sources on those
signals, and get confused by this. If anything we would need to add new
signals.
2015-05-07 14:51:45 +02:00
Sebastian Dröge
d22ec49632 rtpsession: Also report internal sources in on-new-ssrc and on-ssrc-active
Without this it seems impossible for an application to easily get notified
about the internal ssrcs that are created, e.g. sender sources, and also
to know when they are active and produce RTCP packets.

https://bugzilla.gnome.org/show_bug.cgi?id=746747
2015-05-06 11:21:22 +02:00
Sebastian Dröge
9865730cfa rtspsrc: Fix up last commit 2015-05-04 16:50:38 +02:00
Sebastian Dröge
d08f488598 rtspsrc: Only do RTX when using a feedback profile 2015-05-04 16:47:30 +02:00
Sebastian Dröge
9d22ad421b rtpsession: The stats min_interval is in seconds, not nanoseconds
We have to scale it to compare it against our clock times.
2015-05-04 14:12:07 +02:00
Sebastian Dröge
afe1d5a89f rtpsession: Only return TRUE if early feedback was requested already and it's early enough 2015-05-04 14:11:00 +02:00
Luis de Bethencourt
06d1ae313d matroska: remove unused property enum items 2015-04-30 15:43:09 +01:00
Tim-Philipp Müller
377c8405aa qtdemux: fix buffer leak on eos in push mode
Based on patch by Guillaume Desmottes.

scenario: validate.http.playback.seek_with_stop.raw_h264_1_mp4

https://bugzilla.gnome.org/show_bug.cgi?id=748617
2015-04-30 13:35:16 +01:00
Sebastian Dröge
178f0a4522 qtdemux: Check for sizes of the rdrf (redirect) atom before accessing the data and use g_strndup() instead of g_strdup()
Thanks to Ralph Giles for reporting this.
2015-04-29 19:41:29 +02:00
Sebastian Dröge
33693525b9 rtspsrc: Only enable retransmissions if there is retransmission info in the SDP
Otherwise we're going to send early RTCP and NACKs in non-feedback sessions
too, which will confuse servers.

https://bugzilla.gnome.org/show_bug.cgi?id=748627
2015-04-29 15:53:09 +02:00
Guillaume Desmottes
7f4f4131df matroskademux: fix seek event leak
gst_matroska_demux_handle_seek_event() doesn't consume the
event so we have to unref it.

https://bugzilla.gnome.org/show_bug.cgi?id=748584
2015-04-28 19:24:40 +01:00
Sebastian Dröge
9119fbd774 matroska-demux: Send pending tags when adding a new pad
We might've parsed those tags before already and tried to push them to
non-existing pads before. Now let's do it for real.
2015-04-28 15:42:49 +02:00
Sebastian Dröge
73c0c2920f rtpstats: Average RTCP packet size is in bytes, bandwidths in bits
We need to convert the size to bits for our calculations.

https://bugzilla.gnome.org/show_bug.cgi?id=747863
2015-04-27 16:45:40 +02:00
Sebastian Dröge
475b1e607e rtpstats: Use the same lower limit for RTCP bandwidth to stop sending RTCP everywhere
https://bugzilla.gnome.org/show_bug.cgi?id=747863
2015-04-27 16:45:33 +02:00
Sebastian Dröge
7596ed91b8 rtpsession: Use bandwidth calculation by default instead of some arbitrary hardcoded value
https://bugzilla.gnome.org/show_bug.cgi?id=747863
2015-04-27 16:45:25 +02:00
Sebastian Dröge
928cd110bc rtpsession: Bandwidth is supposed to be in bits/s, not bytes/s
https://bugzilla.gnome.org/show_bug.cgi?id=747863
2015-04-27 16:45:14 +02:00
Luis de Bethencourt
9391622579 Rename property enums from ARG_ to PROP_
Property enum items should be named PROP_ for consistency and readability.
2015-04-27 11:22:11 +01:00
Ilya Konstantinov
fd391a5404 rtpjitterbuffer: Fix "stats" property docs
https://bugzilla.gnome.org/show_bug.cgi?id=748436
2015-04-26 21:15:44 +02:00
Tim-Philipp Müller
d753a3eeb1 Remove obsolete Android build cruft
This is not needed any longer.
2015-04-26 17:55:07 +01:00
Thiago Santos
0ade8b813f videocrop: print the property values when set
Instead of printing the currently used values. The log is meant
to show what the properties changed to, not what is being currently
used.
2015-04-24 13:55:51 -03:00
Luis de Bethencourt
671b4d25cd remove unused enum items PROP_LAST
This were probably added to the enums due to cargo cult programming and are
unused. Removing them.
2015-04-24 17:01:12 +01:00
Tim-Philipp Müller
03d3d36053 level: fix infinite loop for very low interval values
https://bugzilla.gnome.org/show_bug.cgi?id=745515
2015-04-24 00:51:29 +01:00
Jesper Larsen
3528046773 rtspsrc: Fix RTCP caps leak
https://bugzilla.gnome.org//show_bug.cgi?id=748353
2015-04-23 14:56:27 +01:00
Sebastian Dröge
edcc5be297 rtpjitterbuffer: When request retransmissions for future packets, consider the packet spacing in the extra delay
We now take the maximum of 2*jitter and 0.5*packet_spacing for the extra
delay. If jitter is very low, this should prevent unnecessary retransmission
requests to some degree.

https://bugzilla.gnome.org/show_bug.cgi?id=748041
2015-04-22 20:27:18 +02:00
Sebastian Dröge
3fe8ceff14 rtpjitterbuffer: Take a running average of the packet spacings instead of just the latest
https://bugzilla.gnome.org/show_bug.cgi?id=748041
2015-04-22 20:25:43 +02:00
Miguel París Díaz
f81c9a9568 rtpjitterbuffer: Add "rtx-next-seqnum" property
If this is set to FALSE, rtpjitterbuffer will not request retransmissions for
future packets based on when they are estimated to arrive.

See also https://bugzilla.gnome.org/show_bug.cgi?id=748041

https://bugzilla.gnome.org/show_bug.cgi?id=739868
2015-04-22 19:51:18 +02:00
Sebastian Dröge
68dfe93463 rtxreceive: Put debug output for retransmission requests at the right place
Before it was only ever printed once for every time a ssrc was associated with
a specific stream.
2015-04-22 19:51:18 +02:00
Luis de Bethencourt
c884a3b3a5 equalizer: fix dynamic changes on bands
When we are in passthrough, the transform function doesn't run and if the
passthrough check is in this function it will never be deactivated. Fix this by
checking directly whenever a gain is changed.

Also set the passthrough to TRUE at init because the gains default to 0, so we
can passthrough until any gain property is changed.

https://bugzilla.gnome.org/show_bug.cgi?id=748068
2015-04-22 10:38:39 +01:00
Vincent Penquerc'h
6e3835594c ac3parse: fix memory leak 2015-04-17 13:33:09 +01:00
Alex O'Konski
fc038f1f4e icydemux: Fix segfault if metadata-interval is 0
Prevents an extra unref of GstBuffer when passing a non-icy stream through
icydemux with metadata-interval set to 0.

Reproducible with:
gst-launch-1.0 filesrc location=~/testsong.mp3 ! \
'application/x-icy,metadata-interval=(int)0' ! icydemux ! decodebin ! wavenc ! \
filesink location=~/testsong.wav

https://bugzilla.gnome.org/show_bug.cgi?id=748024
2015-04-17 10:01:02 +01:00
Ravi Kiran K N
fd6a5a5d90 audiofx: fix typo in example pipelines
Fix typo in example pipelines

https://bugzilla.gnome.org/show_bug.cgi?id=748022
2015-04-17 09:53:46 +01:00
Sebastian Dröge
80268e7d37 rtpsource/rtprtxsend: Also pass correct seqnum-offset and payload to the RTX rtpsource
https://bugzilla.gnome.org/show_bug.cgi?id=747394
2015-04-16 17:33:37 +02:00
Arun Raghavan
26bec72e52 rtpsession: Track RTX ssrc caps
This is needed so that we can generate SR for RTX stream correctly (the
clock rate is required).

https://bugzilla.gnome.org/show_bug.cgi?id=747394
2015-04-16 17:33:37 +02:00
Sebastian Dröge
17c6532b75 rtprtxsend: Copy over timestamps from the orignal buffers to the RTX buffers
https://bugzilla.gnome.org/show_bug.cgi?id=747394
2015-04-16 17:33:37 +02:00
Vincent Penquerc'h
f02ad47998 qtdemux: fix tag list leaks on error paths 2015-04-16 13:10:22 +01:00
Vincent Penquerc'h
765faa306a qtdemux: fix tag list leak on unknown stream type 2015-04-16 13:10:21 +01:00
George Kiagiadakis
97c03449a4 splitmuxsink: do not access property variable without the object lock, use the local stack copy instead 2015-04-15 13:30:19 +02:00
George Kiagiadakis
1954726328 splitmuxsink: add probe on the multiqueue's sink pad instead of the ghost pad
because _release_pad tries to release it from ctx->sinkpad, which is
multiqueue's sink pad, and currently fails because the probe is not
installed there
2015-04-15 13:30:19 +02:00
Sebastian Dröge
caa255d2ed rtprtx*: Fix typos 2015-04-14 19:08:38 +02:00
Sebastian Dröge
bd19b08d6d rtpsession: Not sending early RTCP now because of dithering means we send it with the next compound packet 2015-04-14 18:42:44 +02:00
Sebastian Dröge
4223d0c114 rtpsession: Improve debug output a bit if we can't allow early feedback 2015-04-14 18:42:44 +02:00
Olivier Crête
1394a66e62 rtpvp8depay: When dropping intra packet, request keyframe
https://bugzilla.gnome.org/show_bug.cgi?id=747208
2015-04-13 18:13:35 -06:00
Sebastian Dröge
6c27293ffe rtpjitterbuffer: Change resyncing GST_WARNING to GST_INFO
This also happens in the very beginning when we receive the first packet, a
warning would be very confusing here. In all places where we should warn about
this, we would've printed a warning already before.
2015-04-13 20:25:48 +02:00
Tim-Philipp Müller
b745cb8a47 multifilesink: minor docs improvement 2015-04-13 14:31:17 +01:00
Miguel París Díaz
c4bb6a098b rtpjitterbuffer: Add "rtx-max-retries" property
This property allows to limit the maximum number of retransmission
for a specific packet.

https://bugzilla.gnome.org/show_bug.cgi?id=739868
2015-04-13 09:09:03 +02:00
Miguel París Díaz
05bd708fc5 rtpjitterbuffer: Fix expected_dts calc in calculate_expected
Right above we consider lost_packet packets, each of them having duration,
as lost and triggered their timers immediately. Below we use expected_dts
to schedule retransmission or schedule lost timers for the packets that
come after expected_dts.

As we just triggered lost_packets packets as lost, there's no point in
scheduling new timers for them and we can just skip over all lost packets.

https://bugzilla.gnome.org/show_bug.cgi?id=739868
2015-04-13 09:06:33 +02:00
Sebastian Dröge
1a2f253c3a rtpjitterbuffer: Make the next output buffer discont after resetting the jitterbuffer
Resetting the jitterbuffer drops all packets and other things, and will cause
a discontinuity in the packets received by the depayloaders. They should now
also flush anything they had pending as the new data will start at a different
position.

https://bugzilla.gnome.org/show_bug.cgi?id=739868
2015-04-13 09:05:34 +02:00
Hyunjun Ko
7fbd1b472f qtdemux: Update segment.start after key-unit seek
When doing key uint seek, qtdemux calls gst_qtdemux_adjust_seek
to get proper offset. And then this offset is set to
segment.position and segment.time in gst_qtdemux_perform_seek but
segment.start is not updated.

After that, application sends segment query,
qtdemux sets start and stop to query using gst_segment_to_stream_time. Due
to the wrong value in segment.start, the stop position is smaller than
it should.

https://bugzilla.gnome.org/show_bug.cgi?id=746822
2015-04-10 10:12:50 -03:00
Thiago Santos
39c09284e2 qtmux: remove useless variable do_pts
We always write the CTTS in qtmux. Ideally we only want to do that
for streams that need DTS, it should be present on the track information
rather than be decided based on each buffer
2015-04-10 10:05:24 -03:00
Thiago Santos
5780afe131 qtmux: remove subtraction that makes PTS/DTS start from 0
As qt uses durations, it doesn't matter, only the difference
between consecutive buffers is important. Also, collectpads
already replaces PTS/DTS with the running times for them.
2015-04-10 10:05:24 -03:00
Ravi Kiran K N
d8ebddfaf3 smpte: remove unused fields
Remove the fields - format and fps from smpte
as they are unused.

https://bugzilla.gnome.org/show_bug.cgi?id=747597
2015-04-10 10:23:55 +01:00
Vincent Penquerc'h
a862db33b6 splitmuxsink: fix mutex leak 2015-04-09 13:01:23 +01:00
Jan Schmidt
fe739b7f88 isomp4: Refactor various state variables into a mux_mode var
Instead of checking various state variables around the muxer,
track the current muxing mode in a single 'mux_mode' enum.

Add some implementation notes about the different mux modes
2015-04-09 10:20:06 +10:00
Edward Hervey
5e0329235e rtph263depay: Fix framesize parsing
The string passed to the parsing function only contains a framesize, and
not <pt> + <framesize>

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726416
2015-04-08 11:17:31 +02:00
Vincent Penquerc'h
8cfebfec8c wavparse: clip chunk size above the valid maximum (0x7fffffff)
https://bugzilla.gnome.org/show_bug.cgi?id=722567
2015-04-07 12:12:44 +01:00
Vincent Penquerc'h
3ac119bbe2 wavparse: clip chunk length to available data (when known)
This prevents silly chunk lengths from possibly overflowing
(at least when we know the actual data length).

https://bugzilla.gnome.org/show_bug.cgi?id=722567
2015-04-07 12:12:44 +01:00
Sebastian Dröge
bf95f93c01 qtdemux: Don't accumulate segment bases manually
gst_segment_do_seek() does that for us already, and doing it twice
will break non-flushing seeks in interesting ways. Leftover from 1.0
porting.

Also copy over segment offset and applied_rate, just in case.
2015-04-06 20:17:52 -07:00
Thiago Santos
aeb4d32363 qtdemux: stbl_index is valid from 0 onwards
It indicates the last sample parsed, not the next one to parse.
As it starts in -1, any value from 0 onwards means that it has
some valid data.
2015-04-06 19:29:03 -03:00
Tim-Philipp Müller
2fde2011b2 docs: make GstRTCPSync enum show up in rtpbin docs
https://bugzilla.gnome.org/show_bug.cgi?id=747358
2015-04-05 20:07:19 +01:00
Thiago Santos
cf7d9f676d multifilesink: close files before posting message
Makes sure the files were properly flushed and closed before
the message reaches the application
2015-04-04 11:55:00 -03:00
Thiago Santos
e00f0de4f3 multifilesink: post file message on EOS
When multifilesink is operating in any mode other than one file
per buffer, the last file created won't have a file message posted
as multifilesink doesn't handle the EOS event.

This patch fixes it by using the last position to post a file
message when EOS is received. This should ensure at least the
time related data and the filename are posted to the application
or other elements

https://bugzilla.gnome.org/show_bug.cgi?id=747000
2015-04-04 07:58:44 -03:00
Jan Schmidt
ffa5fce094 qtdemux: Guard against 64-bit overflow
For large-file atoms, guard against overflow in the size field,
which could make us jump backward in the file and cause
infinite loops.
2015-04-03 23:07:07 +11:00
Jan Schmidt
3d59b5f814 isomp4: Make non-seekable downstream an error in normal mode
When not in fast-start or fragmented mode, we need to be able
to rewrite the size of the mdat atom, or else the output just
won't be playable - the mdat placeholder with size == 0 will
cover the rest of the file, including any moov atom we write out.

https://bugzilla.gnome.org/show_bug.cgi?id=708808
2015-04-03 23:07:04 +11:00
Sebastian Rasmussen
cf54d4cc67 rtph263pay/-depay: add framesize SDP attribute
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726416
2015-04-02 19:38:21 -04:00
Sebastian Rasmussen
896fc20806 rtpjpegpay/-depay: Remove incorrectly introduced framesize SDP attribute
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726415
2015-04-02 17:52:41 -04:00
Olivier Crête
d410acf649 rtpvp8depay: Parse width/height/profile from keyframes
This makes it possible to mux the result into a container
such as matroska.

https://bugzilla.gnome.org/show_bug.cgi?id=747208
2015-04-01 19:31:18 -04:00
Jan Schmidt
c0d4986c8d flv: When passing seek event upstream, hold a ref.
In case upstream can't handle the seek, make sure we
keep a ref on the event to attempt to handle it ourselves.
2015-03-31 00:20:48 +11:00
Guillaume Desmottes
592cab1512 matroska: fix GValue leaks when parsing tags
gst_tag_list_add_value() doesn't consume the GValue we pass to it so there is
no point copying it.

https://bugzilla.gnome.org/show_bug.cgi?id=746810
2015-03-30 08:59:36 -03:00
Mark Nauwelaerts
71b0b8d943 qtdemux: resurrect some flow return handling
https://bugzilla.gnome.org/show_bug.cgi?id=744572
2015-03-29 13:58:56 +02:00
Mark Nauwelaerts
33cc1b4854 flvdemux: resurrect some flow return handling
https://bugzilla.gnome.org/show_bug.cgi?id=744572
2015-03-29 13:58:56 +02:00
Mark Nauwelaerts
593cfa086c matroskademux: resurrect some flow return handling
https://bugzilla.gnome.org/show_bug.cgi?id=744572
2015-03-29 13:58:56 +02:00
Thiago Santos
d56b11af56 matroska: store stream tags and push as updated
New tags can be found on different parts of the file, so this patch
keeps the stream taglists around for the life cycle of the pad
and adds those new tags as found. Then a new tag is found, the
pad's is marked with a tags changed flag, making the element push
a new tag event on the next check. Before this, we were sending
only the newly found tags, as the element was losing its taglist
when pushing the event.
2015-03-28 11:20:39 -03:00
Ramiro Polla
7b2b619a8f matroskademux: send global tags incrementally
Instead of sending only new tags once they are found, merge the taglist
and send them incrementally.
2015-03-28 10:24:57 -03:00
Ramiro Polla
af45021036 matroskaparse: send global tags
Global tags are already being read in matroskaparse, but they are not
currently being sent.

This patch makes global tags get sent incrementally whenever new ones
are found.

https://bugzilla.gnome.org/show_bug.cgi?id=746242
2015-03-28 10:24:57 -03:00
Vineeth T M
fb5394dbf0 quarktv: fix "planes" property range, a value of 0 is not allowed
When planes property is set to 0, the pipeline executes in
an infinite loop and never exits. Since planes must never
be 0, set the minimum value in the property description
to 1.

https://bugzilla.gnome.org/show_bug.cgi?id=743906
2015-03-28 11:31:42 +00:00
David Schleef
59756c1898 wavparse: Fix up comments regarding DTS 2015-03-26 16:24:52 -07:00
Nicolas Dufresne
84725d62b5 rtspsrc: Fix segment in TCP mode
It is expected that buffers are time-stamped with running time. Set
a segment accordingly. In this case we pick 0,-1 as this is what udpsrc
would do. Depayloaders will update the segment to reflect the playback
position.

https://bugzilla.gnome.org/show_bug.cgi?id=635701
2015-03-26 17:54:08 -04:00
David Schleef
c3bb399fd3 wavparse: be more strict about typefinding DTS
Code now matches comments.
2015-03-26 12:22:43 -07:00
Nicolas Dufresne
32aed67144 rtspsrc: Remove useless function
This function didn't do anything special, let's not use a function for
that.
2015-03-25 15:28:24 -04:00
Nicolas Dufresne
12762ad1a5 rtpjitter: Account for rtx_retry in overflow check
As rtx_retry is part of the substraction, we need to take it into
account, otherwise we may endup with a big value.
2015-03-25 15:25:56 -04:00
Nicolas Dufresne
8afc8c8f3b rtspsrc: Fix seeking query
The segment start/stop in the query is meant to represent the seekable
portion of the stream. It does not match the segment start/stop. Instead
export 0 to duration.
2015-03-24 16:51:12 -04:00
Sebastian Dröge
ac0141b6a0 flvdemux: Only set caps once if they don't change
Previously we were setting new caps with the same content for every H264 or
AAC codec_data we found in the stream, spamming everything and causing
renegotiations.
2015-03-24 16:18:53 +01:00
Sebastian Dröge
c9b42951fe flvdemux: Don't create AAC/H264 caps without codec_data
Instead delay creating the caps until we read the codec_data from the stream,
or fail if we get normal data before the codec_data.

AAC raw caps and H264 avc caps always need codec_data, setting caps on the pad
without them is going to make negotiation fail most of the time. Even if we
later set new caps with the codec_data, that's usually going to be too late.

https://bugzilla.gnome.org/show_bug.cgi?id=746682
2015-03-24 16:15:04 +01:00
Sebastian Dröge
5e88b53212 flvdemux: Fix indention 2015-03-24 15:39:40 +01:00
Sebastian Dröge
0e3609a6e1 rtpsession: Fix another instance of sticky event misordering warnings
Make sure that the sync_src pad has caps before the segment event.
Otherwise we might get a segment event before caps from the receive
RTCP pad, and then later when receiving RTCP packets will set caps.
This will results in a sticky event misordering warning

This fixes warnings in the rtpaux unit test but also in the
rtpaux and rtx examples in tests/examples/rtp

https://bugzilla.gnome.org/show_bug.cgi?id=746445
2015-03-21 19:30:32 +01:00
Sebastian Dröge
17d90b453f rtpsession: Also start the RTCP send thread when receiving RTP or RTCP
Before we only started it when either:
- there is no send RTP stream
or
- we received an RTP packet for sending

This could mean that if the send RTP pads are connected but never receive any
RTP data, and the same session is also used for receiving RTP/RTCP, we would
never start the RTCP thread and would never send RTCP for the receiving part
of the session.

This can be reproduced with a pipeline like:

gst-launch-1.0 rtpbin name=rtpbin \
udpsrc port=5000 ! "application/x-rtp, media=video, clock-rate=90000, encoding-name=H264" ! rtpbin.recv_rtp_sink_0 \
udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \
rtpbin.send_rtcp_src_0 ! fakesink name=rtcp_fakesink silent=false async=false sync=false \
rtpbin.recv_rtp_src_0_2553225531_96 ! decodebin ! xvimagesink \
fakesrc ! valve drop=true ! rtpbin.send_rtp_sink_0 \
rtpbin.send_rtp_src_0 ! fakesink name=rtp_fakesink silent=false async=false sync=false -v

Before this change the rtcp_fakesink would never send RTCP for the receiving
part of the session (i.e. no receiver reports!), after the change it does.

And before and after this change it would send RTCP for the receiving part of
the session if the sender part was omitted (the last two lines).
2015-03-21 17:38:07 +01:00