Commit graph

14746 commits

Author SHA1 Message Date
Sebastian Dröge
559fd76d7d docs: update to git 2015-12-16 09:35:38 +01:00
Nicolas Dufresne
e7a59d2e08 Revert "alsasrc: Disable HW timestamp"
This reverts commit 3642e9a391.
2015-12-14 13:59:02 -05:00
Xavier Claessens
429860e51f base: Add g_autoptr() support to all types
https://bugzilla.gnome.org/show_bug.cgi?id=754464
2015-12-14 13:39:43 -05:00
Nicolas Dufresne
3642e9a391 alsasrc: Disable HW timestamp
This is a workaround for broken pulse module.
2015-12-14 13:39:02 -05:00
Sebastian Dröge
b0c834df1b rtspconnection: Properly initialize stack-allocated RTSP message to all-zeroes 2015-12-14 19:03:33 +01:00
Evan Callaway
5ac65d9e3a rtspconnection: Use relative URI for non-proxy tunneled requests
Match the section 5.1.2 of the HTTP/1.0 spec by using relative URIs unless we
are using a proxy server. Also, send Host header for compatability with
HTTP/1.1 and some HTTP/1.0 servers.

https://bugzilla.gnome.org/show_bug.cgi?id=758922
2015-12-14 18:21:10 +01:00
Evan Callaway
65c7bd7a2c rtspconnection: Support authentication during tunneling setup
gst_rtsp_connection_connect_with_response accepts a response pointer
which it fills with the response from setup_tunneling if the
connection is configured to be tunneled.  The motivation for this is to
allow the caller to inspect the response header to determine if
additional authentication is required so that the connection can be
retried with the appropriate authentication headers.

The function prototype of gst_rtsp_connection_connect has been
preserved for compatability with existing code and wraps
gst_rtsp_connection_connect_with_response.

https://bugzilla.gnome.org/show_bug.cgi?id=749596
2015-12-14 16:00:45 +01:00
Sebastian Dröge
d6be67265f rtpbasedepayload: Check if the packet loss event actually has timestamp and duration fields
CID 1139615
2015-12-14 13:11:21 +01:00
Wim Taymans
f5a3f70571 audio: adapt API for non-interleaved formats
Allow an array of sample blocks to be passed to the channel mix and
quantizer functions to support non-interleaved formats.
2015-12-14 09:16:08 +01:00
Wim Taymans
aec17c63fd audio-converter: improve API for non-interleaved formats
Make it possible to pass an array of sample blocks when dealing with
non-interleaved formats.
2015-12-14 09:16:08 +01:00
Luis de Bethencourt
055ed65d92 riff: add FourCC aliases
Support media using the aliases defined in http://www.fourcc.org/ that are
exact duplicates of already known codes.
2015-12-12 20:22:44 +00:00
Luis de Bethencourt
98e93ec5ee riff: use defined FourCC
Make gst_riff_create_video_caps() use the FourCC available in riff-ids.h,
like gst_riff_create_audio_caps() does.
2015-12-12 20:22:09 +00:00
Julien Isorce
4f396ae61c videodecoder: add some debug around pool negotiation
It lets us know easily which pool is activated or
inactivated during the negotiation.

https://bugzilla.gnome.org/show_bug.cgi?id=720597
2015-12-11 14:55:46 +00:00
Song Bing
a476145797 video/convertframe: Add crop meta support via videocrop
https://bugzilla.gnome.org/show_bug.cgi?id=759329
2015-12-11 16:14:59 +02:00
Tim-Philipp Müller
f0db396e63 rtpbasedepay: when setting discont flag make sure rtpbuffer is current
Depayloaders will look at rtpbuffer->buffer for the discont flag.
When we set the discont flag on a buffer in the rtp base depayloader
and we have to make the buffer writable, make sure the rtpbuffer
actually contains the newly-flagged buffer, not the original input
buffer. This was introduced with the addition of the process_rtp_packet
vfunc, but would only trigger if the input buffer wasn't flagged
already and was not writable already.
2015-12-11 11:06:35 +00:00
Tim-Philipp Müller
29cd7966b7 tests: rtpbasedepayload: add test for seqnum gap discont setting
The problem was triggered only when the input buffers were not
writable, so add extra ref to test this code path.
2015-12-11 10:40:49 +00:00
Tim-Philipp Müller
86350ff8b7 rtpbasedepay: fix possible refcounting issue when detecting a discont
When we detect a discont and the input buffer isn't already flagged
as discont, handle_buffer() does a gst_buffer_make_writable() on the
input buffer in order to set the flag. This assumed it had ownership
of the input buffer though, which it didn't. This would still work
fine in most scenarios, but could lead to crashes or mini object
unref criticals in some cases when a discont is detected, e.g. when
using pcapparse in front of a depayloader. This problem was
introduced in bc14cdf529.
2015-12-11 10:38:14 +00:00
Wim Taymans
9c2bcd7b76 multisocketsink: add GstNetworkMessage event
Add a property and logic to send a GstNetworkMessage event containing
the message that was received from a client. This can be used to
implement simply bidirectional communication.
2015-12-10 12:44:42 +01:00
Wim Taymans
9aaaa26ff3 multisocketsink: add dispatched event
Add a property and logic to send a GstNetworkMessageDispatched
event upstream to notify that a buffer has been sent. This can be used
to keep track of what client received what buffers.
2015-12-10 12:44:42 +01:00
Wim Taymans
0e1a858d89 socketsrc: handle GstNetworkMessage events
Add a property to handle GstNetworkMessage events. These events contain
a buffer that is sent on the socket to allow for simple bidirectional
communication.
2015-12-10 12:44:42 +01:00
Wim Taymans
5e55968546 audio-convert: improve converter API
Improve the converter API to allow for an max input and output number of
samples and return the number of consumed/produced samples.
2015-12-09 17:16:26 +01:00
Philippe Normand
872f40d7d9 appsrc: duration query support based on the size property
https://bugzilla.gnome.org/show_bug.cgi?id=759126
2015-12-08 12:42:46 +02:00
Nicolas Dufresne
8b49a3f845 Automatic update of common submodule
From b319909 to 86e4663
2015-12-07 09:08:05 -05:00
Wim Taymans
1da5a3ab66 multisocketsink: let downstream know we support metadata
Let downstream know that we support GstNetControlMessage metadata API.
2015-12-04 12:25:11 +01:00
Edward Hervey
d34aaf9e9b videodecoder: Avoid pushing buffers before segment start
In the case where the stream doesn't have a framerate set and the frames
don't have a duration set, we still want to use the clipping path to
make sure we don't push buffers outside of the segment.

The problem was the previous iteration was setting a duration of 2s, which
meant that any buffer which was less than 2s before the segment start would
end up getting pushed.

Instead, use a saner 40ms (25fps single frame duration) to figure out whether
the frame could be within the segment or not
2015-12-03 16:42:50 +01:00
Reynaldo H. Verdejo Pinochet
4ed7b0a0e6 Drop usage of deprecated g-ir-scanner --strip-prefix flag 2015-12-02 20:19:43 -08:00
Tim-Philipp Müller
71505dfa24 decodebin2: fix "Attempt to unlock mutex that was not locked"
Introduced in commit ee44337f, caused the decodebin
test_text_plain_streams unit test to abort.

https://bugzilla.gnome.org/show_bug.cgi?id=752651
2015-12-02 18:16:05 +00:00
Edward Hervey
d292ed48c5 playback: Expose XSUB formats by default
This is a workaround, we should remove this once we have a proper
decoder
2015-12-02 16:37:50 +01:00
Edward Hervey
f9b9472ad4 discoverer: Also consider XSUB as a subtitle format 2015-12-02 16:37:50 +01:00
Edward Hervey
817c780380 pbutils: Add description for XSUB subpicture format 2015-12-02 16:37:50 +01:00
Edward Hervey
27f2328348 riff: 'DXSA' is the same as 'DXSB'
Which is subpicture/x-xsub
2015-12-02 16:37:50 +01:00
Edward Hervey
c79bf13bc2 streamsynchronizer: Rename GstStream => GstSyncStream
Avoid clashes with future GstStream from core
2015-12-02 16:37:41 +01:00
Evan Callaway
e47643122c rtspconnection: Update capitalization of x-sessioncookie
Some servers incorrectly parse header names with strict case-sensitivity.  For
compatibility with these systems change X-Sessioncookie to x-sessioncookie.

https://bugzilla.gnome.org/show_bug.cgi?id=758921
2015-12-02 16:29:53 +02:00
Sebastian Dröge
9e4bf58b8e decodebin: Update buffering messages when removing an element that had buffering pending
Otherwise we'll remove that element while keeping its buffering message in our
list, and because of that never ever report buffering 100% as that element
will always be at a lower percentage.

This fixes e.g. seeking over Period boundaries in DASH and various other
issues when buffering happens between group switches.

Also use a new mutex for protecting the buffering messages. The object lock is
already used by gst_object_has_as_ancestor() and we need to use it now for
checking if the buffering message sender has the to-be-removed element as
ancestor.
2015-12-02 16:16:22 +02:00
Wim Taymans
01f5ca3da8 multisocketsink: keep on reading when we stop sending
When we stop sending because we need more data, still keep a GSource
around to receive data from the clients.
Also handle read and write in the same go.
2015-12-02 10:26:03 +01:00
Sebastian Dröge
2f3eb47a95 audiobasesrc: Post latency message on the bus after set_caps()
The latency is only known once the caps are known, and might change
whenever the caps are changing.

https://bugzilla.gnome.org/show_bug.cgi?id=758911
2015-12-01 19:58:25 +02:00
Michael Olbrich
43155807cd audiobasesink: Post latency message on the bus after set_caps()
Any latency query before this will not get the correct latency so a new
latency query should be triggered once the audio sink know its own latency.

Without this the initial latency query from the pipeline arrives too early
sometimes and the resulting latency is too short.

https://bugzilla.gnome.org/show_bug.cgi?id=758911
2015-12-01 19:58:25 +02:00
Thomas Bluemel
2c62aad159 [PATCH] Fix a race condition accessing the decode_chain field.
Make sure that any access to the GstDecodeBin's decode_chain
field is protected using the EXPOSE_LOCK.  Also add a simple
reference counter to the GstDecodeChain structure so that when
the type_found signal fires it can hold onto the decode chain
even while the EXPOSE_LOCK is not held.  This should fix a
race condition if the type_found signal fires right in the
middle of a state change that messes with the same decode
chain.

https://bugzilla.gnome.org/show_bug.cgi?id=755260
2015-12-01 17:36:31 +00:00
Vincent Penquerc'h
870c6df489 decodebin: early out on pad-added when the pad is inactive
The pad may be recently deactivated if the element is switched
back down very quickly.

https://bugzilla.gnome.org/show_bug.cgi?id=752651
2015-12-01 17:36:31 +00:00
Vincent Penquerc'h
ee44337fc3 decodebin: lock the expose lock around decode_chain use
Helps with a crash in decodebin when quickly switching states.

https://bugzilla.gnome.org/show_bug.cgi?id=752651
2015-12-01 17:36:31 +00:00
Luis de Bethencourt
2a70c86e85 codec-utils: accept wrong version field in OpusHead header
Some Opus files found on the wild have 0 in the version field of the
OpusHead header, instead of the correct value of 1. The files still
play, don't make this error fatal.

https://bugzilla.gnome.org/show_bug.cgi?id=758754
2015-12-01 15:47:35 +00:00
William Manley
aae0dc37c9 allocators: add debug category for fd memory and allocator
Debugging can now be viewed by setting GST_DEBUG=fdmemory:9

https://bugzilla.gnome.org/show_bug.cgi?id=758744
2015-11-27 15:33:47 +00:00
Tim-Philipp Müller
e8a403d181 tests: tags: add unit test for ID3v2 PRIVATE_DATA tag extraction
https://bugzilla.gnome.org/show_bug.cgi?id=730926
2015-11-20 20:20:25 +00:00
Ravi Kiran K N
df5725e683 id3v2frames: Handle private frames
Handle PRIV ID3 tag having owner information (string)
and binary data, add to tag messages list.

https://bugzilla.gnome.org/show_bug.cgi?id=730926
2015-11-20 20:20:18 +00:00
Tim-Philipp Müller
93a92d7f70 tags: id3: make sure to register private-id3v2-frame tag before using it 2015-11-20 19:15:22 +00:00
Ognyan Tonchev
7a702df863 rtspconnection: Add support for parsing custom headers
https://bugzilla.gnome.org/show_bug.cgi?id=758235
2015-11-18 00:15:32 +00:00
Reynaldo H. Verdejo Pinochet
0c95b0a738 Remove unnecessary NULL checks before g_free()
g_free() is NULL-safe
2015-11-17 14:50:27 -08:00
Vineeth TM
5f79ccb420 xvimagesink/ximagesink: Fix structure memory leak
https://bugzilla.gnome.org/show_bug.cgi?id=758204
2015-11-17 00:07:35 -03:00
Luis de Bethencourt
09c881ee14 codec-utils: guint8 can't hold value over 255
channels is a guint8, so the max value is 255 and checking if it value is
> 256 will never be false.

CID 1338687, CID 1338688
2015-11-12 14:39:22 +00:00
Luis de Bethencourt
df16e8dd5a audio-converter: remove unneeded check for unsigned < 0
Commit ff6d1a2a25 changed sample's type from
gint to gsize (and renamed it to in_samples). gsize is an unsigned long,
which means it can never be a negative value and the check making sure that
in_samples is >= 0 is never going to be false. Removing it.

CID 1338689
2015-11-12 14:18:30 +00:00