Commit graph

8347 commits

Author SHA1 Message Date
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
Santiago Carot-Nemesio
22791413f9 rtp: Add Packet Loss Indication (PLI) to statistics
This is helpful to provide statistics in the format defined in
http://w3c.github.io/webrtc-stats/#dictionary-rtcrtpstreamstats-members.

https://bugzilla.gnome.org/show_bug.cgi?id=745587
2015-03-04 12:04:07 +01:00
Nicola Murino
c4e542de69 matroskamux: Remove duration accumulation logic
Duration accumulation can cause rounding errors and generate wrong
duration with different buffers that share the same timestamp.

https://bugzilla.gnome.org/show_bug.cgi?id=745192
2015-03-04 11:37:48 +01:00
Nicola Murino
f727762c1f matroska: Add an helper method to get buffer timestamps
... and replace GST_BUFFER_TIMESTAMP that always return PTS with this method
that return PTS or DTS based on stream type.

https://bugzilla.gnome.org/show_bug.cgi?id=745192
2015-03-04 11:36:24 +01:00
Sebastian Dröge
8984e18ef7 rtpsession: Add explanation why we have space for 32 hash tables
And also create only one, there's no need yet to create all 32 until
we implement RFC2762.
2015-03-04 11:30:43 +01:00
Sebastian Dröge
af2bdd6e15 Revert "rtpsession: Do not use an array of maps if they are not being used"
This reverts commit 1591adf4cd.

https://bugzilla.gnome.org/show_bug.cgi?id=745586#c1:
It's the beginning of an implementation of RFC 2762, which is needed for
large multicast groups. The implementation is not yet complete but why
not leave what is there and implement RFC 2762 instead?
2015-03-04 11:26:57 +01:00
Santiago Carot-Nemesio
1591adf4cd rtpsession: Do not use an array of maps if they are not being used
rtpsession declares an array of maps to store srrcs but only the
the key 0 is being used. This patch replaces the array of maps
for just one map and remove useless parameters in rtpsession

https://bugzilla.gnome.org/show_bug.cgi?id=745586
2015-03-04 11:25:30 +01:00
Jimmy Ohn
42599eab76 avidemux: remove not needed code
In gst_avi_demux_handle_src_query, there is not needed code.
We already check about stream is vbr or not at the upper line.
o, we don't need to check this condition becase stream is not
vbr 100% in this case.

https://bugzilla.gnome.org/show_bug.cgi?id=745276
2015-03-04 10:08:21 +01:00
Matej Knopp
f75e443a7a qtdemux: fix key unit seek
Unlike many other seek flags, the KEY_UNIT seek
flag is not copied over into the GstSegment,
since it's only relevant for the seek itself,
so we need to pass it explicitly to the seek
handler here.

https://bugzilla.gnome.org/show_bug.cgi?id=745339
2015-03-01 13:06:55 +00:00
Nicola Murino
e676b8ba9c matroskamux/demux: initialize dts_only
https://bugzilla.gnome.org/show_bug.cgi?id=745192
2015-02-27 09:56:06 +02:00
Nicola Murino
09b8f0efc3 matroskamux: store DTS for V_MS/VFW/FOURCC streams
https://bugzilla.gnome.org/show_bug.cgi?id=745192
2015-02-27 09:56:06 +02:00
Tim-Philipp Müller
f5b511b42b multifile: attempt to fix docs build issue on build bot 2015-02-26 19:48:33 +00:00
Arun Raghavan
0c06553fb2 interleave: Drop custom latency query handling
This is implemented by the default query handler now.
2015-02-27 00:59:43 +05:30
Arun Raghavan
dbc142afec videomixer: Drop custom latency querying logic
This is now implemented in the default latency query handler.
2015-02-27 00:59:43 +05:30
Sebastian Rasmussen
d331d931db rtpvorbispay: fix payloader description and author e-mail
https://bugzilla.gnome.org/show_bug.cgi?id=745226
2015-02-26 15:57:08 +00:00
Matej Knopp
fa283f407f matroskademux: V_MS/VFW/FOURCC streams have DTS instead of PTS
When such stream is present demuxer should set DTS on buffers instead
of PTS. This is consistent with how VLC and libav/ffmpeg handle VFW
streams.

Sample file
https://s3.amazonaws.com/MatejK/Samples/Matroska-VFW-DTS-Only.mkv

https://bugzilla.gnome.org/show_bug.cgi?id=745192
2015-02-26 11:12:34 +02:00
Krzysztof Kotlenga
e3ca4d1c86 rtspsrc: improve error message when unauthorized
Make use of NOT_AUTHORIZED error code instead of falling back to generic
READ error.

https://bugzilla.gnome.org/show_bug.cgi?id=601733
2015-02-24 11:08:27 +02:00
Thibault Saunier
fa0870658d qtdemux: All segment resulting from a seek should have the same seqnum
https://bugzilla.gnome.org/show_bug.cgi?id=744983
2015-02-23 20:05:20 +01:00
Vincent Penquerc'h
dc73d153cb rtpvp8pay: default encoding name to VP8
https://bugzilla.gnome.org/show_bug.cgi?id=737810
2015-02-19 14:29:02 +00:00
Vincent Penquerc'h
b88ea286d2 rtpvp8pay: make caps writable before truncating them
https://bugzilla.gnome.org/show_bug.cgi?id=737810
2015-02-19 14:06:51 +00:00
Vincent Penquerc'h
b866c989f5 rtpvp8pay: negotiate encoding name
Chrome uses a different one than gstreamer.

https://bugzilla.gnome.org/show_bug.cgi?id=737810
2015-02-19 13:52:29 +00:00
Sebastian Dröge
939a95d44b rtpsession: Send initial events on sync_rtcp pad when using RTP/RTCP muxing
Otherwise we will just send buffers on the pad without any events beforehand
and will get g_warnings() about that.
2015-02-19 13:34:47 +02:00
Thiago Santos
84b7cf6795 qtmux: remove not needed condition
gst_buffer_replace can handle NULL inputs by itself
2015-02-18 10:36:06 -03:00
Thiago Santos
a12e41c106 qtdemux: prefer the tfdt timestamp over the buffer's that is less accurate
The tfdt should be more accurate as the buffer timestamp is provided
by the fragmented format manifest and it might just be an approximation.
2015-02-18 09:57:48 -03:00
Sebastian Dröge
735c6c40f8 rtpjitterbuffer: When resetting the jitterbuffer because of packet discont, don't flush sticky events
We will otherwise flush away STREAM_START, CAPS or SEGMENT events and will
confuse downstream with buffers that come before such events.
2015-02-17 16:57:55 +02:00
Edward Hervey
6798dc7912 isomp4: Redefine gst_isoff_ symbols to gst_isoff_qt_
We need different symbol names, because these symbols are also present
in the fragmented plugin ... which will cause conflicts when doing
static linking
2015-02-17 12:31:06 +01:00
Luis de Bethencourt
ea1d67abe3 goom2k1: use fractional part of float division 2015-02-16 14:31:05 +00:00
Luis de Bethencourt
4af5a2b760 splitmuxsin: remove dead code
Every instance of goto beach has buf_info equal NULL. Don't check
for a condition that never happens.

CID #1268399
2015-02-16 13:59:17 +00:00
Nicolas Dufresne
b8142bde07 spectrum: Fix min and max for bands property
The number of FFTs is calculated with the following formula:

  guint nfft = 2 * bands - 2;

nfft is passed to gst_fft_f32_new() as the len argument and is of type
unsigned integer. This method required that len is at leas 1, then
maximum G_MAXINT, as other values would be negative. If we extrapolate
from the formula above it means we need "bands" to be between 2 and
((guint)G_MAXINT + 2) / 2).

https://bugzilla.gnome.org/show_bug.cgi?id=744213
2015-02-15 21:34:28 -05:00
Thiago Santos
afa5481c50 qtdemux: do not use sparse streams in push-based seeking
Using the sparse streams can make the push-based seeking return
too far in the stream. It also can lead to issues as the
sparse streams will be ignored when restarting playback and,
 if the sparse stream is the one that has the earliest sample,
it will confuse qtdemux's offsets as one stream will have
an earlier offset than the demuxer's one which might lead to
early EOS.

https://bugzilla.gnome.org/show_bug.cgi?id=742661
2015-02-14 11:36:11 -03:00
Tim-Philipp Müller
3f5b690e78 splitmuxsink: flag as sink from the start 2015-02-13 20:40:48 +00:00
Philippe Normand
3a9b0188cd qtdemux: Initial 'sidx' atom parsing support
Parse the 'sidx' atom and update the total duration according to the
parser result. The isoff parser code is imported from
gst-plugins-bad's dashdemux and a gst_isoff_sidx_parser_add_data()
function was factored out of the gst_isoff_sidx_parser_add_buffer()
function.

https://bugzilla.gnome.org/show_bug.cgi?id=743578
2015-02-12 14:23:21 -03:00
Jan Schmidt
2e00311fe1 flvdemux: Use gst_video_guess_framerate()
Use gst_video_guess_framerate() from libgstvideo to guess
sensible common framerates where possible from the
floating point fps in the stream.
2015-02-12 23:38:47 +11:00
Sebastian Dröge
f4b5107796 Improve and fix LATENCY query handling
This now follows the design docs everywhere, especially the maximum latency
handling.

https://bugzilla.gnome.org/show_bug.cgi?id=744106
2015-02-11 13:53:02 +01:00
Sebastian Dröge
b79eff7f9b rtpsession: Handle first RTCP packet and early feedback correctly
According to RFC 4585 section 3.5.3 step 1 we are not allowed to send
an early RTCP packet for the very first one. It must be a regular one.

Also make sure to not use last_rtcp_send_time in any calculations until
we actually sent an RTCP packet already. In specific this means that we
must not use it for forward reconsideration of the current RTCP send time.
Instead we don't do any forward reconsideration for the first RTCP packet.
2015-02-11 10:32:46 +01:00
Wim Taymans
009a62fddb rtph263depay: fix compilation with gcc 5.0 2015-02-10 18:54:24 +01:00
Tim-Philipp Müller
90badeebad splitmuxsink: fix example pipeline properly
x264enc might not have a max-key-int property, but it
has a key-int-max property...
2015-02-10 16:00:07 +00:00
Luis de Bethencourt
102ae8511a splitmux: fix typo 2015-02-10 14:57:55 +00:00
Luis de Bethencourt
12aa2428e0 splitmux: update example pipeline
Element x264enc doesn't have a max-key-int property
2015-02-10 14:56:23 +00:00
Luis de Bethencourt
0373fd8f65 splitmux: fix memory leak
If execution goes to the beach in line 981, buf_info goes out of scope without
the memory being free'd. Handle this case.

CID #1268403
2015-02-10 13:33:09 +00:00
Tim-Philipp Müller
603c1d71a1 rtspsrc: fix awkward if clause 2015-02-08 12:03:10 +00:00
Jan Schmidt
8ceb58122e splitmux: Add unit test for file splitting
Add a unit test for file splitting, and fix the leaks in the
splitmuxsink it found
2015-02-07 03:58:30 +11:00
Luis de Bethencourt
eb975ce880 wavparse: fix which stop variable is used in assignment
Assignment is done to variable segment.stop when the intention was to assign to
local variable stop. Instead of overwriting it, the value is now clamped and
segment.stop is set to it soon after.

CID #1265773
2015-02-06 14:46:14 +00:00
Jan Schmidt
aa4c29c5d6 splitmux: Fix memory leaks until the test valgrinds clean 2015-02-07 00:19:36 +11:00
Jan Schmidt
ace6be8abb splitmux: Handle early EOS during part preparation
Handle the case where a short file reaches EOS while we're still
waiting for no-more-pads, and make sure we continue to the internal
READY state for real playback to work properly later.
2015-02-06 06:42:17 +11:00
Jan Schmidt
5e2214d309 splitmux: Implement new elements for splitting files at mux level.
Implement 2 new elements - splitmuxsink and splitmuxsrc.

splitmuxsink is a bin which wraps a muxer and takes 1 video stream,
plus audio/subtitle streams, and starts a new file
whenever necessary to avoid overrunning a threshold of either bytes
or time. New files are started at a keyframe, and corresponding audio
and subtitle streams are split at packet boundaries to match
video GOP timestamps.

splitmuxsrc is a corresponding source element which handles
the splitmux:// URL and plays back all component files,
reconstructing the original elementary streams as it goes.
2015-02-06 04:26:59 +11:00
Thiago Santos
a6d73797d0 rtph264depay: prevent trying to get 0 bytes from adapter
This causes an assertion and would lead to getting a NULL instead
of a buffer. Without proper checking this would easily lead to
a segfault

https://bugzilla.gnome.org/show_bug.cgi?id=737199
2015-02-04 21:37:50 -03:00
Jan Schmidt
a3059bec1f qtdemux: Simple implementation of GST_SEGMENT_FLAG_TRICKMODE_KEY_UNITS
When the trickmode key-units flag is set on the segment, simply skip
any sample on a video stream that isn't a keyframe
2015-02-04 21:58:31 +11:00
Wim Taymans
852c040c89 rtspsrc: fix container handling
We detect a container correctly now so we need to revert the weird
check there was before.
Use gst_rtspsrc_stream_push_event() to push the caps event on the
right pad.

See https://bugzilla.gnome.org/show_bug.cgi?id=739391
2015-02-03 17:39:10 +01:00
Thiago Santos
7772a25fdc matroskamux: store and write stream tags
Separate global from stream tags storage and write them to the
appropriate tags entry in the output
2015-02-02 20:07:13 -03:00
Thiago Santos
75dee31b0d qtdemux: parse stream tags
Keep global and stream tags separately and parse the udta node
that can be found under the trak atom. The udta will contain
stream specific tags and will be pushed as such

https://bugzilla.gnome.org/show_bug.cgi?id=692473
2015-02-02 14:05:51 -03:00
Thiago Santos
e52b2cb2cf qtmux: store stream and container tags separately
Tags received via events, when marked as stream tags, will
be stored on that stream's trak atom instead of being stored
in the main tags atom. This allows the resulting file to have
global and stream tags stored.

https://bugzilla.gnome.org/show_bug.cgi?id=692473
2015-01-31 17:23:01 -03:00
Thiago Santos
6321cdedb3 qtmux: refactor tags functions to accomodata UDTA at trak level
Refactor the functions that were bound to the 'moov' atom to
directly pass the desired 'udta' that should receive the tags.
This allows the tags to be written to 'udta' at the 'moov' or
the 'trak' level, creating tags that are for the container or
for a stream only.

https://bugzilla.gnome.org/show_bug.cgi?id=692473
2015-01-31 17:22:57 -03:00
Thiago Santos
f0fde8be88 qtmux: map application name to _swr tag
It refers to the application name and version used to create the
file

https://bugzilla.gnome.org/show_bug.cgi?id=692473
2015-01-31 17:22:44 -03:00
Jan Schmidt
4a77c8a84f matroska: Fix seeking past the end of the file in reverse mode.
Snap to the end of the file when seeking past the end in reverse mode,
and also fix GST_SEEK_TYPE_END and GST_SEEK_TYPE_NONE handling
for the stop position by always seeking on a segment in stream time
2015-01-31 06:15:44 +11:00
Sebastian Dröge
075eb10e65 rtpsession: Fix signal name
This wasn't meant to be pushed at all yet, but now that it's there
already it won't hurt to make it correct at least.
2015-01-30 18:22:31 +01:00
Sebastian Dröge
ec99bbb5e1 rtpstats: Fix typo in documentation 2015-01-30 16:56:35 +01:00
Sebastian Dröge
77511b156e rtpsession: Add new on-receiving-rtcp signal
This will be emitted whenever an RTCP packet is received. Different to
on-feedback-rtcp, this signal gets every complete RTCP packet and not
just the individual feedback packets.
2015-01-30 16:50:36 +01:00
Thiago Santos
9a9d4eccea qtdemux: simplify segment.base math
Remove a fix for heavily edited files added for fixing
https://bugzilla.gnome.org/show_bug.cgi?id=345830 to work
with seeks and proper gaps playback. The fix was replaced
for a more general solution that bases on using previous
segment's duration, just like it works for media segments
playback.

https://bugzilla.gnome.org/show_bug.cgi?id=743518
2015-01-28 15:20:58 -03:00
Luis de Bethencourt
5ff1229754 videomixer: update orc files 2015-01-27 14:00:35 +00:00
Thiago Santos
2586a219f6 qtdemux: Fix data dropping for fragmented streams
For fragmented streams with extra data at the end of the mdat
qtdemux was not dropping those bytes and would try to use
that extra data as the beginning of a new atom, causing the
stream to fail.

https://bugzilla.gnome.org/show_bug.cgi?id=743407
2015-01-27 08:54:19 -03:00
Sebastian Dröge
e4ed852041 rtpsession: Deprecate rtcp-immediate-feedback-threshold property
It had no effect since quite some time and also is not needed in general,
especially not to switch between immediate feedback mode and early feedback
mode. The latest understanding of the RFC is that from the endpoint point of
view, both modes are exactly the same. RTCP is only allowed to use the
bandwidth as given by the RFC constraints, as such it is only ever possible
to schedule a RTCP packet early but it's against the RFC to schedule more RTCP
packets.

The difference between immediate feedback mode and early feedback mode is that
the former guarantees that an RTCP packet can be sent for every event
"immediately", which means that the bandwidth calculations from the RFC have
resulted in an RTCP scheduling interval that is small enough. Early feedback
mode on the other hand means that we can schedule some packets early to make
that happen, but it's not guaranteed at all that it's possible to schedule
an RTCP packet per event (i.e. they need to be accumulated or dropped).
2015-01-26 18:49:31 +01:00
Sebastian Dröge
b07b7736b3 rtpsession: Delay the next regular RTCP packet after early RTCP
This is required to not exceed the short term average RTCP bitrate when
using early feedback as compared to without early feedback.
2015-01-26 18:49:31 +01:00
Sebastian Dröge
bc9111a03d rtpsession: Add new send-rtcp-full signal
This indicates with a boolean return value if scheduling a new RTCP packet
within the requested delay was possible. Otherwise it behaves exactly like
send-rtcp. The only reason for adding a new signal is ABI compatibility.
2015-01-26 18:49:31 +01:00
Luis de Bethencourt
1e15808563 matroskademux: remove unnecessary check
No matter if gst_matroska_read_common_parse_index_cuetrack () returns that the
flow is OK or not, the check there will be a break from the switch. Removing the
check since the outcome is the same.

CID #1265762
2015-01-23 17:35:51 +00:00
Edward Hervey
932b32bb6e matroskamux: Avoid using freed variable
the name variable might have been attributed to pad_name, make sure we
free it only *after* pad_name has been used.

Coverity CID : 1265774
2015-01-23 15:16:25 +01:00
Edward Hervey
8abfd9d720 avimux: Avoid using freed variable
the name variable might have been attributed to pad_name, make sure we
free it only *after* pad_name has been used.

Coverity CID : 1265775
2015-01-23 15:15:07 +01:00
Sebastian Dröge
60e2d0c84f rtpsession: Fix indention 2015-01-22 11:03:25 +01:00
Edward Hervey
7203c4751c qtdemux_dump: Bypass even more code if debugging is disabled
And avoid using variables that won't exist when debugging is disabled
2015-01-21 17:36:26 +01:00
Edward Hervey
906f4c4360 qtdemux: Only traverse/dump nodes if guaranteed to be used
__gst_debug_min is the "global" lowest debug level set. There's no
guarantee the qtdemux debug category is actually set at that level.
2015-01-21 15:32:01 +01:00
Edward Hervey
9fa85f72e1 matroska: Avoid debugging below category threshold
This part alone was what made the matroska thread take a full core
on an android phone ...
2015-01-21 15:26:41 +01:00
Sebastian Dröge
d5aab81a77 Constify some static arrays everywhere 2015-01-21 09:55:53 +01:00
Vincent Penquerc'h
d854cfff9d qtdemux: fix deadlock seeking in files without seek entries
A mutex unlock was missing.

https://bugzilla.gnome.org/show_bug.cgi?id=739975
2015-01-19 17:49:54 +00:00
Vincent Penquerc'h
84c44fceac videomixer: fix illegal memory access in blend function with negative ypos
https://bugzilla.gnome.org/show_bug.cgi?id=741115
2015-01-19 12:34:25 +00:00
Sebastian Dröge
dc2251a664 qtmux: Add support for v210 2015-01-13 19:05:40 +01:00
Sebastian Dröge
b7134435ee qtdemux: v210 is v210, not UYVY and yuv2 is YUY2, not I420
Also add a few other raw video formats we support: v308, v216
and add comments for a few others we don't support yet.

https://developer.apple.com/library/mac/technotes/tn2162/
2015-01-13 19:05:40 +01:00
Thiago Santos
3e0be85840 qtdemux: fix stream time conversion
Use the right macro to convert to the correct scale or the
segment information will be wrong

https://bugzilla.gnome.org/show_bug.cgi?id=742572
2015-01-09 11:40:40 -03:00
Matej Knopp
ff5b235c32 ac3parse: request at least 8 bytes to properly parse header
https://bugzilla.gnome.org/show_bug.cgi?id=742325
2015-01-08 14:45:23 +01:00
Michael Smith
e8f3d596bc wavparse: skip an additional uninteresting chunk type before the fmt chunk. 2015-01-07 16:20:03 -08:00
Luis de Bethencourt
42535107ca audiodynamic: assert func_index is inside bounds
Bringing back the check removed in the previous commit but have that check be a
g_assert. Changing the function to static void since return can never be False,
because audio format will never be unkown.
2015-01-07 18:16:12 +00:00
Luis de Bethencourt
1db92a91de audiodynamic: remove always-true conditional
func_index is set by the sum of three ternary operators which add, 0:4, 0:2,
and 1:0. Minimum value would be 0+0+0=0, and maximum would be 4+2+1=7.
The conditional checking if func_index is >= 0 and < 8 will always be true.
Removing it.

CID 1226442
2015-01-07 17:31:39 +00:00
Sebastian Dröge
87c8c163a8 rtpjitterbuffer: If we get a gap with a buffer without DTS, error out
We (currently?) can't really handle gaps between RTP packets if they're not
properly timestamped. The current code would go into calculations with
GST_CLOCK_TIME_NONE and then cause assertions everywhere. It's probably
better to error out cleanly instead.
2015-01-07 18:05:18 +01:00
Aleix Conchillo Flaqué
07c5d1820a rtspsrc: set PLAYING state after configuring caps
We set to PLAYING after we have configured the caps, otherwise we
might end up calling request_key (with SRTP) while caps are still
being configured, ending in a crash.

https://bugzilla.gnome.org/show_bug.cgi?id=740505
2014-12-31 12:49:11 +00:00
Sebastian Dröge
67d4b85d6a matroskademux: Improve detection of being stuck at the same offset
Only error out if we read from the same position again and got the
same length. Just the same position is not necessarily enough.
2014-12-29 15:35:19 +01:00
Sebastian Dröge
e596a3b6a7 matroskademux: Don't get stuck at the same offset when searching for clusters
This could happen if there is an invalid cluster with size 0, and in that
case just error out instead of looping forever.
2014-12-29 15:02:52 +01:00
Tim-Philipp Müller
aa94fc6beb qtmux: fix ALAC muxing
Actually copy the codec data instead of copying nothing
and then bombing out because there's no data.

Fixes: gst-launch-1.0 audiotestsrc ! avenc_alac ! qtmux ! fakesink

https://bugzilla.gnome.org/show_bug.cgi?id=741783
2014-12-25 21:37:49 +00:00
Tim-Philipp Müller
c62209d050 rtpptdemux: just drop invalid rtp packets instead of erroring out
Apparently linphone sends an invalid RTP packet as very
first packet. We want to ignore that instead of erroring
out (same for any other invalid packets really).

https://bugzilla.gnome.org/show_bug.cgi?id=741398
2014-12-25 15:48:04 +00:00