Commit graph

4563 commits

Author SHA1 Message Date
Stefan Sauer
7bbfa39ada audioconvert: fix passthrough operation
We did not take the sample size into account. Rearrange the tests to have more
conversion test and an extra test case for passthrough operations.

Fixes #759890
2015-12-29 14:40:32 +01:00
Reynaldo H. Verdejo Pinochet
e61f5b2138 videoblend: special case 1x1 src dims on increment computation
Fix crash with 1x1 overlay pixmap

https://bugzilla.gnome.org/show_bug.cgi?id=757290
2015-12-28 14:16:41 -08:00
Stefan Sauer
0bd3f818bb audio-converter: code cleanup
Rename samples to num_samples, since we also have samples in chain, but that is
the data pointer. Always use gzize for num_samples. Make the log output a bit
more homogenous.
2015-12-27 19:25:20 +01:00
Sebastian Dröge
3459bd6854 audio: Fix some documentation warnings
Remove/rename function parameters and skip some functions that can't
be used by bindings as they are now.
2015-12-26 09:43:56 +01:00
Sebastian Dröge
3ba59f0b62 videoaffinetransformmeta: Add (transfer none) annotation for return value 2015-12-26 09:43:51 +01:00
Sebastian Dröge
7fddeaa878 pbutils: Link to libgstbase for bytewriter and adapter 2015-12-24 13:59:52 +01:00
Thibault Saunier
512ac3ea72 encodebin: Implement an encoding profile serialization format
https://bugzilla.gnome.org/show_bug.cgi?id=759356
2015-12-24 09:52:53 +01:00
Kazunori Kobayashi
d43f1b2a5a appsrc: Clear is_eos flag when receiving the flush-stop event
The EOS event can be propagated to the downstream elements when
is_eos flag remains set even after leaving the flushing state.
This fix allows this element to normally restart the streaming
after receiving the flush event by clearing the is_eos flag.

https://bugzilla.gnome.org/show_bug.cgi?id=759110
2015-12-19 11:35:39 +01:00
Wim Taymans
08734e7598 audio-converter: rework the main processing loop
Rework the main processing loop. We now create an audio processing
chain from small core functions. This is very similar to how the
video-converter core works and allows us to statically calculate an
optimal allocation strategy for all possible combinations of operations.
Make sure we support non-interleaved data everywhere.
Add functions to calculate in and out frames and latency.
2015-12-16 11:13:15 +01:00
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
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
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
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
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
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
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
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
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
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
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
Vineeth TM
3f099e3c29 pbutils:encoding-target: Fix string memory leak
https://bugzilla.gnome.org/show_bug.cgi?id=757926
2015-11-11 15:40:52 +01:00
Vineeth TM
b61e1465b7 audio-quantize: Fix dither_buffer memory leak
https://bugzilla.gnome.org/show_bug.cgi?id=757928
2015-11-11 15:01:08 +01:00
Matthew Waters
0b98ed32ce videometa: add GstVideoAffineTransformationMeta
Adds a simple 4x4 affine transformations meta for passing arbitrary
transformations on buffers.

Based on patch by Matthieu Bouron

https://bugzilla.gnome.org/show_bug.cgi?id=731791
2015-11-11 00:19:25 +11:00
Wim Taymans
ff6d1a2a25 audio-converter: add output size argument
Make it possible to have a different number of output samples than input
samples when we, for example, want to add resampling later.
2015-11-10 09:53:59 +01:00
Thibault Saunier
629b63d1f2 discoverer: Check API arguments and assert if needed 2015-11-07 00:46:47 +01:00
Wim Taymans
30977cf1a5 audio-converter: require interleaved samples and no resampling
We can't yet do resampling or anything other than interleaved audio.
2015-11-06 18:00:41 +01:00
Wim Taymans
7abed02858 audio: update ORC dist files 2015-11-06 17:54:21 +01:00
Wim Taymans
e3f0f3b91e audio-converter: move audio converter to audio libs
Move the audio-converter helper to the audio library.
2015-11-06 17:53:22 +01:00
Wim Taymans
dfa25a40fc audio-channel-mix: move channel mixer to audio libs
Move the channel mixer code to the audio library
2015-11-06 17:39:33 +01:00
Wim Taymans
b8bea9d8be audio: add debug categories 2015-11-06 17:29:22 +01:00