Commit graph

15427 commits

Author SHA1 Message Date
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
c16c381a89 tests: Update mp4 mux test for mdat placeholder change
The mp4 muxer now writes a place-holder mdat as a free
atom followed by a 0-byte mdat that covers the rest of the
file, making it possible to rewrite it as 64-bit, or leave
it as-is if nothing else is written afterward
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
Stefan Sauer
d7794a6f61 Automatic update of common submodule
From 21ba2e5 to d37af32
2015-06-07 23:06:20 +02:00
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
Stefan Sauer
7c4d8230ce Automatic update of common submodule
From c408583 to 21ba2e5
2015-06-07 17:32:01 +02:00
Stefan Sauer
cd5fe9d3a2 docs: remove variables that we define in the snippet from common
This is syncing our Makefile.am with upstream gtkdoc.
2015-06-07 17:18:49 +02:00
Stefan Sauer
31510e2883 Automatic update of common submodule
From d676993 to c408583
2015-06-07 17:16:19 +02:00
Sebastian Dröge
6c48ce3a28 Back to development 2015-06-07 16:44:37 +02:00
Sebastian Dröge
a7faa3e0a2 Release 1.5.1 2015-06-07 10:46:34 +02:00
Sebastian Dröge
5c93af74a6 Update .po files 2015-06-07 10:38:28 +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
39c0137ea1 po: Update translations 2015-06-07 09:35:38 +02:00
Nicolas Dufresne
0196fdb5ec v4l2: Don't warn when optional CID are not implement
gst_v4l2_get_attributre() shall only be used when the CID is expected
to be supported. Otherwise, we get unwanted warning posted to the bus.
2015-06-05 15:34:12 -04:00
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
Edward Hervey
d524439b35 check: Use GST_CHECK_MAIN () macro everywhere
Makes source code smaller, and ensures we go through common initialization
path (like the one that sets up XML unit test output ...)
2015-06-02 16:27:24 +02:00
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
7bd1cfa197 examples: Set RTP profile to AVPF for rtpaux examples
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