Commit graph

8333 commits

Author SHA1 Message Date
Nicolas Dufresne
717265ebfb flmux: Make sure best_time is initialized 2015-06-12 17:45:23 -04:00
Sebastian Dröge
dc513eb949 rtpbin/session: Add new ntp-time-source property and deprecate use-pipeline-clock property
The new property allows to select the time source that should be used for the
NTP time in RTCP packets. By default it will continue to calculate the NTP
timestamp (1900 epoch) based on the realtime clock. Alternatively it can use
the UNIX timestamp (1970 epoch), the pipeline's running time or the pipeline's
clock time. The latter is especially useful for synchronizing multiple
receivers if all of them share the same clock.

If use-pipeline-clock is set to TRUE, it will override the ntp-time-source
setting and continue to use the running time plus 70 years. This is only kept
for backwards compatibility.
2015-06-12 23:35:42 +02:00
Nicolas Dufresne
135e516730 qtdemux: Adjust segment according to ctts offset
In presence of a CTTS, the segment start/stop must be offset so
the segment start/stop include the PTS. This is needed since the
PTS cannot be negative in this format. This fixes issues where the
running time of the first buffer isn't at the start.

https://bugzilla.gnome.org/show_bug.cgi?id=740575
2015-06-12 17:18:24 -04:00
Nicolas Dufresne
12181efddc qtmux: Handle DTS with negative running time
As QT works with duration, simply bring back first DTS to 0 and shift
forward the PTS of the same amount.

https://bugzilla.gnome.org/show_bug.cgi?id=740575
2015-06-12 17:18:24 -04:00
Nicolas Dufresne
2274ca7d07 flvmux: Add negative runtime DTS support
This is done by using new feature of the CollectPad clip function
which sets the DTS as a gint64 in the collected data. It also simplify
the code a bit.

https://bugzilla.gnome.org/show_bug.cgi?id=740575
2015-06-12 17:18:24 -04:00
Sebastian Dröge
37e3ca1447 rtpbin: Rename some variables and debug output to make more sense
Local and remote were mixed up in a few places, and the time we store here is
not UNIX time (1970 epoch), but NTP time (1900 epoch) in nanoseconds.
2015-06-12 23:07:27 +02:00
Jan Schmidt
0c46c5c3e2 matroska-demux: Actually set detected 3D info into output caps.
Use the information read from the StereoMode info
to configure multiview-mode and multiview-flags in the
video caps.
2015-06-12 01:57:36 +10:00
Jan Schmidt
3f39d06338 splitmuxsink: Take released-but-not-yet-output bytes into account
When deciding whether it's time to switch to a new file, take into
account data that's been released for pushing, but hasn't yet
been pushed - because downstream is slow or the threads haven't been
scheduled.

Fixes a race in the unit test and probably in practice - sometimes
failing to switch when it should for an extra GOP or two.

Also fix a problem in splitmuxsrc where playback sometimes
stalls at startup if types are found too quickly.

https://bugzilla.gnome.org/show_bug.cgi?id=750747
2015-06-12 01:57:36 +10:00
Thiago Santos
03f1a2ea67 atoms: remove custom gst_buffer_new function in favor of core version
Remove a custom specialized version of gst_buffer_new_wrapped by
using gst_buffer_new_wrapped_full inside a macro to simplify
parameters and give it a more meaningful name.
It is only used to create temporary buffers to have its data copied.
2015-06-11 01:11:31 -03:00
Thiago Santos
1596972674 atoms: simplify free form data atoms creation
Avoid creating an intermediary buffer or memory area just
to copy into an atom's data area.
2015-06-11 01:11:31 -03:00
Thiago Santos
ab18f5035c qtmux: add AC-3 muxing support
Adds AC-3 muxing support. It is defined for mp4 and 3gp formats.

One extra feature that was added was the ability to add extension
atoms after set_caps as the AC-3 extension atom needs some data
that has to be extracted from the stream itself and is not
present on caps.
2015-06-11 01:11:31 -03:00
Thiago Santos
674e0cc2df qtmux: remove unused type MP4S 2015-06-11 01:11:31 -03:00
Thiago Santos
f83fd7a88f qtmux: remove duplicate attribute value set
It is also set a few lines below
2015-06-11 01:11:18 -03:00
Jan Schmidt
ec5bc9dccb matroska: Implement basic stereoscopic video support
Implement support for the packed video formats WebM
uses, not all the values that Matroska might use.

In practice, it's really hard to find any samples in the
wild of any.

Supported in both the muxer and demuxer.
2015-06-11 12:11:42 +10:00
Jan Schmidt
fff76157d8 qtdemux: Add basic support for MPEG-A stereoscopic video
The MPEG-A format provides an extension to the ISO base media
file format to store stereoscopic content encoded with different
codecs like H.264 and MPEG-4:2. The stereo video media information(svmi)
atom declares the presence and storage method for the video.

Stereo video information for MPEG-A can also be supplied through
the 'stvi' atom (ref: ISO/IEC_14496-12, ISO/IEC_23000-11), which
is not implemented in this patch.

Also missing is support for stereo video encoded as separate video tracks
for now.

Based on a patch by Sreerenj Balachandran <sreerenj.balachandran@intel.com>

https://bugzilla.gnome.org/show_bug.cgi?id=611157
2015-06-11 12:11:42 +10:00
Sebastian Dröge
dc059efa60 rtp: Use GST_BUFFER_PTS() instead of GST_BUFFER_TIMESTAMP()
The mix between all these in the RTP code is confusing, let's try to be
consistent.
2015-06-10 14:34:47 +02:00
Ilya Konstantinov
c7e168ec70 rtpmanager: clarify negative lost packets in stats
Also:
- Move notes on units before field documentation.
- Unify documentation style.

https://bugzilla.gnome.org/show_bug.cgi?id=750653
2015-06-10 14:10:52 +02:00
Vineeth TM
720ff75c72 qtdemux: fix reverse playback
When performing seek, segment->start is being updated with desired_offset,
but in case of reverse playback segment->start should be 0 and
segment->stop should be updated with desired offset.

https://bugzilla.gnome.org/show_bug.cgi?id=750675
2015-06-10 10:41:13 +02:00
Xavier Claessens
b0b3e8e2cc rtspsrc: Add a GTlsInteraction property
It can be used for TLS client authentication.

https://bugzilla.gnome.org/show_bug.cgi?id=750471
2015-06-09 20:03:18 -04:00
Ilya Konstantinov
0a578c235a rtpmanager: document units of stats and arguments
Also, minor spelling and style corrections.

https://bugzilla.gnome.org/show_bug.cgi?id=750653
2015-06-09 18:21:59 +02:00
Luis de Bethencourt
e56ef6bcf0 goom: possible uninitialized variables warning
Build fails with the latest snapshot of gcc-4.9 because param1 and param2 might
possibly be used uninitialized. They are set depending on the cases of a switch
statement and the compiler sees this as not a complete guarantee.
Set them to 0 if the switch statement falls down to the default case.

https://bugzilla.gnome.org/show_bug.cgi?id=750566#c6
2015-06-08 23:06:39 +01:00
Chris Clayton
e29f231e5d rtpvp8depay: potential access beyond end of array
Compiling (with gcc-4.9-20150603) produces an error because of an access beyond
the end of an array. This patch fixes the error by initializing the loop
control/array index variable (i) to 1 and returning i - 1 when a match is found.
Also, because the values stored in the array increase in value as the index
increases, the >= test unnecessary, so it is removed.
2015-06-08 20:16:20 +01:00
Jan Schmidt
d78502deb1 splitmuxsink: Don't accumulate more than 2 GOPs
Don't allow large amounts of data to queue up - we only need
the GOP we're writing, and the GOP we're accumulating.
2015-06-08 18:58:43 +10:00
Jan Schmidt
23d610140d isomp4: fsync after sending updates in robust mode
Use the new GstBuffer SYNC_AFTER flag to trigger an fsync
after updating the moov or mdat atom, and after updating the free
atom to make it visible.
2015-06-08 14:49:11 +10:00
Jan Schmidt
3e17cd8acb isomp4: Only set moov header into streamheader at EOS
Only update the moov header into the caps if it's the finalised
moov at EOS time. Avoids posting a bogus moov at startup and
repeated updates in robust-recording mode
2015-06-08 14:49:11 +10:00
Jan Schmidt
1d058c7d8a isomp4: Implement robust muxing using ping-pong strategy
Implement a robust recording mode, where the output
file is always in a playable state, seeking and rewriting
the moov header at a configurable interval. Rewriting
moov is done using reserved space at the start of
the file, and a ping-pong strategy where the moov
is replaced atomically so it's never invalid.

Track when tags have actually changed, and don't write them into
the moov unless they've changed. Clear any existing tags when
re-writing them, so we can do progressive moov updating in robust
recording mode.

Write placeholder mdat as a free atom plus a 32-bit mdat
with '0' size, which means "rest of the file" in the spec.

Re-write it later to a full 64-bit extended size atom if needed.
2015-06-08 14:49:11 +10:00
Jan Schmidt
3d7b343525 isomp4: Update edit list when re-writing moov
Correctly update any edit lists each time the moov is recalculated,
updating existing table entries if they already exist instead of just
adding new ones.
2015-06-08 14:16:36 +10:00
Jan Schmidt
0c1bcc629d isomp4: Remove an extra bracket in a comment. 2015-06-08 14:16:36 +10:00
Jan Schmidt
94e113c6c6 splitmuxsrc: Protect total_duration state variable with the object lock.
Prevent deadlocks from downstream querying duration from the streaming thread.
2015-06-08 14:16:36 +10:00
Luis de Bethencourt
0b8c7ab797 goom: clean dereferences of private structure
https://bugzilla.gnome.org/show_bug.cgi?id=742875
2015-06-07 19:24:20 +01:00
Luis de Bethencourt
fce8e5fb26 goom2k1: clean dereferences of private structure
https://bugzilla.gnome.org/show_bug.cgi?id=742875
2015-06-07 19:20:49 +01:00
Sebastian Dröge
a7faa3e0a2 Release 1.5.1 2015-06-07 10:46:34 +02:00
Sebastian Dröge
b549ebd066 rtpsession: Override the SSRC from the packets' SSRC if none was given via caps or property 2015-06-07 10:33:27 +02:00
Sebastian Dröge
d650a310da rtpsession: Only suggest our internal ssrc if it's not a random one and was selected as internal ssrc
https://bugzilla.gnome.org/show_bug.cgi?id=749581
2015-06-05 16:45:54 +02:00
Vineeth TM
0e5631c5c0 interleave: error when channel-positions-from-input=False
self->channels is being incremented only when
channel-positions-from-input is set as TRUE. So in case of FALSE
self->func is not set and hence creating assertion error.
Hence removing the condition to increment self->channels.

https://bugzilla.gnome.org/show_bug.cgi?id=744211
2015-06-05 08:48:25 -03:00
Sebastian Dröge
8f5bdf9690 rtpjitterbuffer: Add support for receiving reduced size RTCP
It worked before but gave warnings, now we just ignore RTCP
packets that don't start with a SR. As all we're interested
in here are SRs.
2015-06-05 10:33:11 +02:00
Jose Antonio Santos Cadenas
f563176349 rtpssrcdemux: Add support for reduce size rtcp
According to RFC 5506, reduce size packages can be sent, this
packages may not be compound, so we need to add support for
getting ssrc from other types of packages.

https://bugzilla.gnome.org/show_bug.cgi?id=750327
2015-06-05 10:30:15 +02:00
Jose Antonio Santos Cadenas
f8f23bbf5d rtpsession: Add support for receiving reduced size rtcp
See RFC 5506

https://bugzilla.gnome.org/show_bug.cgi?id=750332
2015-06-05 10:24:17 +02:00
Sebastian Dröge
ec82eba96b aacparse: Add support for channel configurations 11, 12 and 14 and 7 actually has 8 channels
ISO/IEC 14496-3:2009/PDAM 4 added 11, 12 and 14.
2015-06-04 16:09:41 +02:00
Nicolas Dufresne
3ab70e4677 asteriskh263: Un-rank clashing depayloader
This depayloader clash with the standard one for H263p. It produces an
H263p stream with a modified header. It uses encoding-name that is the
same as H263p (H263-1998) though the resulting ES is not decodable or
parsable in GStreamer, making it unsuable in dynamic pipeline. This
patch unrank this specialized depayloader since it can only be used in
custom pipeline.

https://bugzilla.gnome.org/show_bug.cgi?id=739935
2015-06-03 08:57:57 -04:00
Luis de Bethencourt
ffe7507512 goom2k1: remove variables not needed anymore
https://bugzilla.gnome.org/show_bug.cgi?id=742875
2015-06-02 18:09:48 +01:00
Luis de Bethencourt
8756b6a9d4 goom2k1: rebase to use the audiovisualizer class
Rebase to have goom2k1 using the common GstAudioVisualizer class

https://bugzilla.gnome.org/show_bug.cgi?id=742875
2015-06-02 18:02:08 +01:00
Luis de Bethencourt
89903bf66a goom: rebase to use the audiovisualizer class 2015-06-02 17:47:57 +01:00
Sebastian Dröge
647eefea67 rtpsession: Only schedule a timer when we actually have to send RTCP
Otherwise we will have 10s-100s of thread wakeups in feedback profiles, create
RTCP packets, etc. just to suppress them in 99% of the cases (i.e. if no
feedback is actually pending and no regular RTCP has to be sent).

This improves CPU usage and battery life quite a lot.

https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Sebastian Dröge
8ada98964d rtpsession: Remove useless goto
https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Sebastian Dröge
0a7823b30f rtspsrc: Set RTP profile on the rtpsession objects
https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Sebastian Dröge
506a8a8857 rtpbin: Add rtp-profile property for setting the default profile of newly created sessions
https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Sebastian Dröge
0f7e80ed59 rtpsession: Only put RRs and full SDES into regular RTCP packets
If we may suppress the packet due to the rules of RFC4585 (i.e. when
below the t-rr-int), we can send a smaller RTCP packet without RRs
and full SDES. In theory we could even send a minimal RTCP packet
according to RFC5506, but we don't support that yet.

https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Sebastian Dröge
6f830e5bd5 rtpsession: Keep track of tp/tn and t_rr_last separately
Otherwise we can't properly schedule RTCP in feedback profiles as we need to
distinguish the time when we last checked for sending RTCP (tp) but might have
suppressed it, and the time when we last actually sent a non-early RTCP
packet.

This together with the other changes should now properly implement RTCP
scheduling according to RFC4585, and especially allow us to send feedback
packets a lot if needed but only send regular RTCP packets every once in a
while.

https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Sebastian Dröge
3122ef4ae3 rtpsession: Add property for selecting RTP profile (AVP/AVPF/etc)
And modify our RTCP scheduling algorithm accordingly. We now can send more
RTCP packets if needed for feedback, but will throttle full RTCP packets by
rtcp-min-interval (t-rr-int from RFC4585).

In non-feedback mode, rtcp-min-interval is Tmin from RFC3550, which is
statically set to 1s or 0s by RFC4585. Tmin defines how often we should
send RTCP packets at most.

https://bugzilla.gnome.org/show_bug.cgi?id=746543
2015-06-02 11:38:15 +02:00
Olivier Crête
8fd3e0e125 mulawdec: Let baseclass estimate bitrate
This makes playback directly from a file work with the right caps.
2015-05-30 17:41:44 -04:00
Tim-Philipp Müller
2e5df10ed9 dynudpsink: keep GCancellable fd around instead of re-creating it constantly
And create it only when starting the element.
2015-05-27 17:08:47 +01:00
Tim-Philipp Müller
b33d30621c udpsink, multiudpsink: keep GCancellable fd around instead of re-creating it constantly
Otherwise we constantly create/close event file descriptors,
every time we call g_socket_condition_timed_wait() or
g_socket_send_message(s)(), i.e. a lot. Which is not
particularly good for performance.

Can't create GCancellable in ::start() here because it's used
in client_new() which may be called via the add-client action
signal which may be called before the element is up and running.
2015-05-27 17:08:47 +01:00
Tim-Philipp Müller
11bb21f3c2 udpsrc: keep GCancellable fd around instead of re-creating it constantly
Otherwise we constantly create/close event file descriptors,
every single time we call g_socket_condition_timed_wait() or
g_socket_receive_message(), i.e. twice per packet received!
This was not particularly good for performance.

Also only create GCancellable on start-up.
2015-05-27 17:08:47 +01:00
Luis de Bethencourt
6d06a74f7f matroska: overwritten value assignment
curpos is set and immediately after, set again. Remove the redundant
assignment.

https://bugzilla.gnome.org/show_bug.cgi?id=749909
2015-05-27 16:56:15 +01:00
Tim-Philipp Müller
80998dadba rtpvrawdepay: don't shadow existing outbuf variable
And fix unref of the wrong one which will contain NULL
in an error code path.
2015-05-25 16:16:47 +01:00
Tim-Philipp Müller
2aafb3951d rtpvrawdepay: map/unmap output frame only once, not for every input packet
Map output buffer after creating it and keep it mapped
until we're done with it instead of mapping/unmapping
it for every single input buffer.
2015-05-25 16:16:42 +01:00
Thiago Santos
d03b9513f1 qtdemux: remove fixme from 2006
It has been verified by use over time.
2015-05-25 08:47:47 -03:00
Thiago Santos
fc0a184592 qtdemux: fix reverse playback of fragmented media
qtdemux creates a samples array and gets the timestamps for buffers by
accumulating their durations. When doing reverse playback of fragments,
accumulating samples will lead to wrong timestamps as the timestamps
should go decreasing from fragment to fragment and the accumulation
will produce wrong results.

In this case, when receiving a discont for fragmented reverse playback,
the previous samples information should be flushed before new data
is processed.
2015-05-25 08:46:18 -03:00
Jimmy Ohn
d3997773fc splitfilesrc: Implement binary search in find_part_for_offset
Implement binary search using gst_util_array_binary_search

https://bugzilla.gnome.org/show_bug.cgi?id=749690
2015-05-25 14:23:32 +10:00
Sebastian Dröge
565cd49643 rtpsession: Don't crash if we receive FIR/PLI from a source we don't know 2015-05-21 13:26:53 +03:00
Santiago Carot-Nemesio
2fb1fe2ee3 rtpsession: Fix collection of statistics
Stats should be collected on the media rtp source not in the
sender one.

https://bugzilla.gnome.org/show_bug.cgi?id=749669
2015-05-21 12:56:12 +03:00
Edward Hervey
27c91bc881 multifilesink: Add a new max-duration file switching mode
This new mode ensures that files will never exceed a certain duration
based on incoming buffer PTS (and duration if present)

Note:
* You need timestamped buffers (duh). If some of the incoming buffers don't
  have PTS, then it will just accept them in the current file
2015-05-20 15:50:07 +02:00
Edward Hervey
f1ceaab02f multifilesink: streamline the file-switch code a bit
Use the same functions regardless of the mode we are using
2015-05-20 15:50:07 +02:00
Edward Hervey
db0abbd531 multifilesink: add "aggregate-gops" property to process GOPs as a whole
This property can be used in combination with next-file=max-size
(and perhaps a future next-file=max-duration) to make sure that
each file part starts cleanly with a key frame and the appropriate headers.

In order for this property to work correctly, upstream elements should make
sure than any headers that need to be written in a standalone file are:
1) in the streamheader caps field
2) and/or in the stream as one or more buffers marked with GST_BUFFER_FLAG_HEADER
   that are just before the keyframe buffer

This is useful for MPEG-TS/MPEG-PS file segmenting in
combination with mpegtsmux or mpegpsmux.

Original patch by: Tim-Philipp Müller <tim@centricular.com>
2015-05-20 15:49:57 +02:00
Sebastian Dröge
9b14170355 rtspsrc: Use single-include header for the RTSP library 2015-05-20 16:37:55 +03:00
Tim-Philipp Müller
f54110fd3e udp: don't use soon-to-be-deprecated g_cancellable_reset()
From the API documentation: "Note that it is generally not
a good idea to reuse an existing cancellable for more
operations after it has been cancelled once, as this
function might tempt you to do. The recommended practice
is to drop the reference to a cancellable after cancelling
it, and let it die with the outstanding async operations.
You should create a fresh cancellable for further async
operations."

https://bugzilla.gnome.org/show_bug.cgi?id=739132
2015-05-19 19:00:20 +01:00
Stefan Sauer
168881a186 Revert "doc: Workaround gtkdoc issue"
This reverts commit 1797c8f8b1.

This is fixed by the gtk-doc 1.23 release.
<para> cannot contain <refsect2>:
http://www.docbook.org/tdg/en/html/para.html
http://www.docbook.org/tdg/en/html/refsect2.html
2015-05-18 20:13:01 +02:00
Nicola Murino
5e226d63f9 rtpg726pay: fix caps leak
https://bugzilla.gnome.org/show_bug.cgi?id=749544
2015-05-18 17:40:55 +01:00
Nicola Murino
335afc982b rtpg726depay: don't leak input buffer
https://bugzilla.gnome.org/show_bug.cgi?id=749543
2015-05-18 17:40:39 +01:00
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
Sebastian Dröge
1018aacb35 rtprtxsend: Add support for buffer lists 2015-03-19 11:54:37 +01:00
Sebastian Dröge
57ff27f8c8 rtprtxqueue: Implement support for buffer lists 2015-03-19 11:54:37 +01:00
Nicolas Dufresne
1c27002ebd rtspsrc: Improve trace readability
Change the command number into strings.
2015-03-18 17:32:36 -04:00
Jan Alexander Steffens (heftig)
be8e3196a3 flvdemux: Don't repeatedly warn after no_more_pads (v2)
This can get rather spammy for such a high log level.
Only warn once per stream.

https://bugzilla.gnome.org/show_bug.cgi?id=746274
2015-03-16 12:01:43 +00:00
Jan Alexander Steffens (heftig)
ac8a272381 flvdemux: Introduce constant for no-more-pads threshold
https://bugzilla.gnome.org/show_bug.cgi?id=746274
2015-03-16 12:01:43 +00:00
Jan Alexander Steffens (heftig)
f2a1f74cec flvdemux: Fix warning to contain 'video'
https://bugzilla.gnome.org/show_bug.cgi?id=746274
2015-03-16 12:01:43 +00:00
Nicola Murino
bb3d82ef04 matroskademux: for dts only stream set pts=dts for intra only formats
https://bugzilla.gnome.org/show_bug.cgi?id=745192
2015-03-15 14:28:36 +00:00
Ramiro Polla
0fad053497 matroskademux: fix sending of tags
* Fix critical when new tags are found after segment event has already
  been sent.
* Send global tags before stream tags.
* Split sending of tags out of gst_matroska_demux_send_event() into its
  own function.

https://bugzilla.gnome.org/show_bug.cgi?id=745973
2015-03-14 18:17:48 +00:00
Ramiro Polla
90be7b4e1e rtspsrc: properly escape percent sign in documentation 2015-03-14 14:22:39 +00:00
Ramiro Polla
63944753b0 rtpdtmfmux: properly escape percent sign in documentation 2015-03-14 14:22:26 +00:00
Tim-Philipp Müller
3c595f308a multiudpsink: fix crash with GST_DEBUG enabled
g_inet_socket_address_get_address() does not give
us a ref to the address, so don't unref it.
2015-03-13 18:38:42 +00:00
Sebastian Dröge
7b90bf3215 level: Don't read over the end of the input memory
Previously we advanced the in_data pointer by bps for every channel, and then
later again for block_size*bps. This caused us to be one sample further than
expected if an input buffer covered two analysis frames. And in the end lead
to completely bogus values reported by level.

https://bugzilla.gnome.org/show_bug.cgi?id=746065
2015-03-12 13:51:56 +00:00
Tim-Philipp Müller
c4fa54da17 Fix double semicolons 2015-03-10 09:31:20 +00:00
Jan Schmidt
d441140cd6 splitmux: Shut down element before downward state change
Make sure the state change won't hang trying to shut down pads
by making sure the streaming has stopped before chaining up.
2015-03-10 15:49:33 +11:00
Luis de Bethencourt
823194284c rtph264depay: remove unused value
CID #1226474
2015-03-09 16:22:33 +00:00
Luis de Bethencourt
5cd293fe76 rtph263pay: fix leak
CID 1212156
2015-03-09 16:17:45 +00:00
Luis de Bethencourt
e87113781a rtph263pay: remove uneeded variable
We just need to save the ebit information in case there is an error decoding.
2015-03-09 16:17:45 +00:00
Luis de Bethencourt
db3ade5bfb matroska: error mode if can't push buffer
If gst_pad_push() fails, inform and return flow error.
2015-03-09 12:51:21 +00:00
Luis de Bethencourt
f494da89b4 matroska: unused value
Value set in ret will be overwritten just before exiting the function.

CID #1226469
2015-03-09 12:13:40 +00:00
Sebastian Dröge
9e934d076b rtpjitterbuffer: Drop packets with sequence numbers before the seqnum-base
These are outside the expected range of sequence numbers and should be
clipped, especially for RTSP they might belong to packets from before a seek
or a previous stream in general.
2015-03-09 11:10:35 +01:00
Linus Svensson
398296d978 rtspsrc: Don't include payload type in the caps for framesize
When the sdp media attribute framesize are converted to caps
the <payload> should not be included.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
2015-03-09 10:18:35 +01:00
Sebastian Dröge
38bf3d3808 rtpjitterbuffer: Don't forget to unlock the mutex when receiving GAPs in TCP streams 2015-03-09 10:05:14 +01:00
Mark Nauwelaerts
d0587467fc avidemux: resurrect some flow return handling 2015-03-07 20:22:33 +01:00
Nicolas Huet
5ead23a14a aacparse: fix LOAS parsing issue
Fix missing index in syncword searching

https://bugzilla.gnome.org/show_bug.cgi?id=745585
2015-03-06 14:34:08 -03:00
Jan Schmidt
b0ce43cde3 splitmuxsink: Protect property variables with the object lock.
Use the object lock instead of the splitmux lock to protect
internal property variables, so they're not locked when
switching to a new file.

https://bugzilla.gnome.org/show_bug.cgi?id=744420
2015-03-07 00:55:47 +11:00
Sebastian Dröge
c34a7cb90d rtspsrc: Fix handling of interleaved (TCP) streams
We need to set up the transport in any case, not just if we have a container
stream or a non-interleaved stream. Only if we have an interleaved stream and
are retrying, we should not set up the stream again.

https://bugzilla.gnome.org/show_bug.cgi?id=745599
2015-03-05 12:15:04 +01:00
Sebastian Dröge
b4aaa11f97 rtspsrc: Don't unref caps we don't own 2015-03-05 09:56:37 +01:00
Sebastian Dröge
297d808acc rtspsrc: Push RTCP caps on the RTCP pads
Otherwise we will get not-negotiated later from rtpbin, and will never be able
to send RTCP packets back to the server. Note that error flow returns from the
RTCP pads are ignored, that's why it didn't fail more visible before.
2015-03-05 09:47:29 +01:00
Sebastian Dröge
788074733c rtspsrc: Make sure to send SEGMENT events on all pads 2015-03-05 09:47:29 +01:00
Santiago Carot-Nemesio
e05378ec16 rtp: Add Full Intra Request (FIR) packets to statistics
https://bugzilla.gnome.org/show_bug.cgi?id=745587
2015-03-04 12:04:40 +01:00