Commit graph

15109 commits

Author SHA1 Message Date
Reynaldo H. Verdejo Pinochet
71c2ddd07d Remove unnecessary NULL checks before g_free()
g_free() is NULL-safe
2015-11-18 16:05:42 -08:00
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
Luis de Bethencourt
6463ff198e opusenc: avoid potential overflow expression
The result of the two expressions will be promoted to guint64 anyway,
perform all the arithmetic in 64 bits to avoid potential overflows.

CID 1338690, CID 1338691
2015-11-12 12:29:33 +00:00
Vineeth TM
b0b0536f91 tests:video: Fix overlay rectangle and buffer leak
Created overlay rectangle is not being freed in video tests
pix2 buffer is being created and not freed

https://bugzilla.gnome.org/show_bug.cgi?id=757927
2015-11-11 15:47:24 +01:00
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
Jan Schmidt
797a0ca376 vorbisdec: Re-init on new caps
If we get new input caps, then reset the decoder
ready for new headers and fresh data. Makes
chained oggs work when reusing the decoder.
2015-11-11 01:33:55 +11:00
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
Edward Hervey
d0eface01c decodebin: Properly deactivate ghostpads
Just setting the ghostpad as flushing wasn't enough. It needs to be
consistent on the internal proxypad also, otherwise you end up in
situations where:
* a pending buffer on the target pad triggers the sticky event
  propagation
* the default implementation sees that the proxypad is not flushing,
  so it tries to push it to the other pad (the actual ghostpad)
* the ghostpad is flushing, so returns FALSE
* the push_event function sees that pushing the event failed...
* ... and pending buffer push returns GST_FLOW_ERROR, instead of
  GST_FLOW_FLUSHING

By using gst_pad_set_active(FALSE), we ensure that both the ghostpad
and the proxypad are flushing/deactivated. The situation above will
no longer occur, and a GST_FLOW_FLUSHING will be returned.
2015-11-06 19:38:13 +01:00
Tim-Philipp Müller
d2e210bbea audioconvert: fix build
Don't include file that is no longer generated, and remove some
files that are no longer needed because they have moved into the
lib. Fixes distcheck.
2015-11-06 18:12:28 +00:00
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
Wim Taymans
268ed5dd6f channelmix: don't limit channelpositions
Don't set a limit on the channel positions, just like the metadata.
2015-11-06 16:42:35 +01:00
Wim Taymans
9fbe0386d0 channelmix: simplify API a little
Remove the format and layout from the mix_samples function and use the
format when creating the channel mixer object. Also use a flag to handle
the unlikely case of non-interleaved samples like we do elsewhere.
2015-11-06 16:03:20 +01:00
Wim Taymans
7f5104f52f channelmix: GstChannel -> GstAudioChannel
Rename GstChannel to GstAudioChannel
2015-11-06 15:50:34 +01:00
Wim Taymans
59db8ce542 audio-quantize: update docs
Update docs
Add another flag for the quantizer
2015-11-06 13:02:19 +01:00
Wim Taymans
1635bc0a45 audioconvert: cleanups and add some docs
Add docs for the internal audioconvert object before moving it to the
audio library.
Remove get_sizes and implement the trivial logic in the element.
Remove some unused orc functions
2015-11-06 12:46:36 +01:00
Wim Taymans
4ae24dcb25 defs: update defs 2015-11-06 12:46:12 +01:00
Wim Taymans
dfbeb78342 audio: update orc files 2015-11-06 12:37:14 +01:00
Wim Taymans
c36ac3ce45 audioconvert: move audio quantize code to libs
Move the audio quantize code from audioconvert to the audio library.
work on making an audio converter helper function similar to the video
converter.
Fold fastrandom directly into the quantizer, add some ORC code to
optimize this later.
2015-11-06 12:10:48 +01:00
Wim Taymans
a7789854d5 audio-channels: rename get_default_mask
Rename _get_default_mask() to _get_fallback_mask() to make it more
clear that the function only provides a fallback if nothing else can be
done. Also clarify this in the documentation.

API: gst_audio_channel_get_fallback_mask()
2015-11-05 12:50:18 +01:00
Sebastian Dröge
dd741e6412 opusdec: Update sink pad templates
We always require the channel-mapping-field. If it's 0 we require nothing
else, otherwise we need channels, stream-count and coupled count to be
available.
2015-11-05 12:11:19 +01:00
Thibault Saunier
9c7d3c8ab2 volume: Do not try to get binding value array if we are not processing any sample
In some conditions we might process empty buffers, calling
gst_control_binding_get_value_array in that case will lead
to the assertion:

  (lt-ges-launch-1.0:18859): GStreamer-CRITICAL **: gst_control_binding_get_value_array: assertion 'values' failed
2015-11-05 11:44:31 +01:00
Wim Taymans
f86ed8cdf6 audio-channels: make method to get default channel-mask
Add a new method to get the default channel-mask.
Use the new method on audiodecoder and audioconvert.

API: gst_audio_channel_get_default_mask()
2015-11-05 10:52:53 +01:00
Andreas Frisch
4a85438e5b tests: Add a test for video blending over transparent frames
And fix the test_overlay_blend test where we blend over a
transparent frame and where expecting wrong results

https://bugzilla.gnome.org/show_bug.cgi?id=681447
2015-11-04 22:17:35 +01:00
Arnaud Vrac
dfe250d17d video: blend using OVER operation
Also support all premultiplied/non-premultiplied source/destination
configurations

https://bugzilla.gnome.org/show_bug.cgi?id=681447
2015-11-04 21:58:32 +01:00
Sebastian Dröge
387839c57e opus: Remove invalid unit test
Opus headers should never be in-band, so don't test for correct
handling of that.
2015-11-04 00:14:13 +02:00
Sebastian Dröge
85984fa946 opusenc: Create an empty taglist if there is none
There always have to be 2 buffers in the streamheaders, even if
the comment buffer is basically empty.
2015-11-04 00:14:13 +02:00
Sebastian Dröge
4ca84a9b1a opus: Add proper support for multichannel audio
https://bugzilla.gnome.org/show_bug.cgi?id=757152
2015-11-03 20:35:41 +02:00
Sebastian Dröge
fc475ce01a opusdec: Handle GstAudioClippingMeta instead of the pre-skip field in the OpusHead
oggdemux is outputting the meta now, and only outputs if it should really
apply to the current buffer. Previously we would skip N samples also if we
started the decoder in the middle of the stream.

https://bugzilla.gnome.org/show_bug.cgi?id=757153
2015-11-03 20:35:41 +02:00
Sebastian Dröge
328f9088f3 opusenc: Add GstAudioClippingMeta to buffers that need to be clipped
https://bugzilla.gnome.org/show_bug.cgi?id=757153
2015-11-03 20:35:41 +02:00
Sebastian Dröge
6b751360ae opusenc: Disable granule position calculations by the base class
It is doing the wrong thing because of the Opus pre-skip: while the timestamps
are shifted by the pre-skip, the granule positions are not shifted.

oggmux is doing the right thing here already.

https://bugzilla.gnome.org/show_bug.cgi?id=757153
2015-11-03 20:35:41 +02:00
Sebastian Dröge
7773e1eb58 opusenc: Add some FIXME comments about calculating padding with LPC
https://bugzilla.gnome.org/show_bug.cgi?id=757153
2015-11-03 20:35:41 +02:00
Sebastian Dröge
4df2ffaad6 opusenc: Encode exactly the amount of samples we got as input and put correct timestamps on it
The first frame has lookahead less samples, the last frame might have some
padding or we might have to encode another frame of silence to get all our
input into the encoded data.

This is because of a) the lookahead at the beginning of the encoding, which
shifts all data by that amount of samples and b) the padding needed to fill
the very last frame completely.

Ideally we would use LPC to calculate something better than silence for the
padding to make the encoding as smooth as possible.

With this we get exactly the same amount of samples again in an
opusenc ! opusdec pipeline.

https://bugzilla.gnome.org/show_bug.cgi?id=757153
2015-11-03 20:35:41 +02:00
Sebastian Dröge
6ffb90e037 opusenc: Put lookahead/pre-skip into the OpusHead header
https://bugzilla.gnome.org/show_bug.cgi?id=757153
2015-11-03 20:35:41 +02:00
Sebastian Dröge
cab8671f0c oggdemux: Create full Opus caps with all fields
https://bugzilla.gnome.org/show_bug.cgi?id=757152
2015-11-03 20:35:33 +02:00
Sebastian Dröge
bcd7b2fff2 codec-utils: Add utilities for Opus caps and the OpusHead header
https://bugzilla.gnome.org/show_bug.cgi?id=757152
2015-11-03 20:35:33 +02:00
Sebastian Dröge
0fa8d284c7 oggmux: Use GstAudioClippingMeta for Opus for accurate end clipping
... instead of relying on the segment. For the clipping at the start we assume
a proper value in the OpusHead, as generated by opusparse or opusenc.

Transmuxing in general is not guaranteed to produce the correct values, or
even have a OpusHead (e.g. when having RTP input).

https://bugzilla.gnome.org/show_bug.cgi?id=757153
2015-11-03 20:35:33 +02:00
Sebastian Dröge
a135868262 oggdemux: Add GstAudioClippingMeta for Opus for accurate start/end clipping
https://bugzilla.gnome.org/show_bug.cgi?id=757153
2015-11-03 20:35:33 +02:00
Sebastian Dröge
35ea6fdddf audio: Add GstAudioClippingMeta for specifying clipping on encoded audio buffers
https://bugzilla.gnome.org/show_bug.cgi?id=757153
2015-11-03 20:35:33 +02:00
Sebastian Dröge
8a3be7323a oggdemux: Allow start clipping for Opus
The granulepos does not have the pre-skip subtracted while timestamps do,
and the last granulepos will be shorter by the number of samples that should
be dropped because of padding in the end.

As such, extrapolating the granule of the beginning of the first frame will
lead to a negative value, which is not a problem but intentional.

https://bugzilla.gnome.org/show_bug.cgi?id=757153
2015-11-03 20:35:33 +02:00