Commit graph

4556 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
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
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
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
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
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
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
Tim-Philipp Müller
1f2fdd3789 audio: update disted orc backup files 2015-11-03 16:38:09 +00:00
Luis de Bethencourt
94a7f9fc4e audioclock: use GST_STIME_FORMAT for GstClockTimeDiff
GST_STIME_FORMAT is more appropriate for GstClockTimeDiff since it can
handle negative values better.

https://bugzilla.gnome.org/show_bug.cgi?id=757480
2015-11-03 14:08:29 +00:00
Luis de Bethencourt
227f1d1e0f videodecoder: Print GstClockTimeDiff as a signed integer in debug logs 2015-11-03 13:44:39 +00:00
Wim Taymans
801f7ca464 audio-format: add TRUNCATE_RANGE flag
Add a TRUNCATE_RANGE flag for unpack functions to fill the least
significate bits with 0 (as did the old code). Also add functions
that don't truncate. Use the TRUNC flag in audioconvert for
backwards compatibility for now.
2015-11-03 12:12:08 +01:00
Wim Taymans
914aa4aed1 audiopack: improve pack functions
Avoid shifts by using convh functions.
2015-11-03 12:12:08 +01:00
Luis de Bethencourt
fe62e797d5 audiobasesink: use GST_STIME_ARGS for GstClockTimeDiff
No need to use G_GINT64_FORMAT for potentially negative values of
GstClockTimeDiff. Since 1.6 these can be handled with GST_STIME_ARGS.
Plus it creates more readable values in the logs.

https://bugzilla.gnome.org/show_bug.cgi?id=757480
2015-11-02 17:35:20 +00:00
Sebastian Dröge
443171bb4c audio: Fix parameters to gst_buffer_get_audio_downmix_meta() in macro 2015-11-02 17:35:45 +02:00
Luis de Bethencourt
b81b3f07ec videodecoder: subtract time difference with GST_CLOCK_DIFF
To ensure the subtraction of two GstClockTime values (which are guint64)
can be negative. Use GST_CLOCK_DIFF which returns a gint64.

CID 1338049
2015-11-02 12:09:45 +00:00
Thibault Saunier
a7123ebb58 encoding-profile: Do not force user to provide an encoding profile name
And use the profile called `default` if none provided.
2015-11-02 11:35:55 +01:00
Thibault Saunier
83fa06aab5 encoding-target: Do not unconditionally break when searching for a target
Otherwise the loop is useless!

Fixes CID 1338051
2015-11-02 11:31:34 +01:00
Sebastian Dröge
736a27fe1e audiofilter: Clip input buffers to the segment before handling them
https://bugzilla.gnome.org/show_bug.cgi?id=757068
2015-11-02 10:20:37 +02:00
Sebastian Dröge
1da79c76a7 videodecoder: Print another time difference as a signed integer instead of a huge unsigned one 2015-10-29 16:52:49 +02:00
Sebastian Dröge
f17758d9e3 videodecoder: Print GstClockTimeDiff as a signed integer in debug logs 2015-10-29 16:01:26 +02:00
Thibault Saunier
2e20f3ba4f encoding-target: Add a GST_ENCODING_TARGET_PATH envvar to find target files 2015-10-28 16:07:44 +01:00
Thibault Saunier
96f69fa998 encoding-target: Allow having encoding target without a category set
There was already some code to handle that, but the support was not
complete in those code paths.
2015-10-28 16:07:44 +01:00
Thibault Saunier
0256381f6f encoding-target: Create directory before trying to save encoding targets 2015-10-28 16:07:44 +01:00
Thibault Saunier
db272cf9cb encoding-profile: Allow specifying the target category in the serialized encoding target 2015-10-28 16:07:44 +01:00
Pavel Bludov
8051434c03 video: overlay-composition: fix rectangle and composition cast macros
Closing parenthesis was missing in two cases.

https://bugzilla.gnome.org/show_bug.cgi?id=756893
2015-10-22 09:24:01 +01:00
Sebastian Dröge
5bdeed32bc rtp: GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING is Since 1.6.1 2015-10-20 12:10:52 +03:00
eunhae choi
e98b96247f audiobasesink: fix issue about eos handling during flushing
If the flush-start is arrived during _eos_wait() in basesink,
the 'eos' flag is overwritten to TRUE after exiting the _eos_wait().
To resolve the overwritten issue,
the subclass doing the _eos_wait() call should return the right value.
If the eos flag is set to TRUE again, it will cause error(enter the eos flow)
of the following state changing from PAUSED to PLAYING in basesink.

https://bugzilla.gnome.org/show_bug.cgi?id=754980
2015-10-19 12:12:12 -03:00
Sebastian Dröge
df0d5aa4d5 video: Add out annotations to the out parameters of gst_video_calculate_display_ratio()
https://bugzilla.gnome.org/show_bug.cgi?id=754567
2015-10-17 20:37:22 +03:00
Stian Selnes
0a668c1866 rtpbuffer: Add map flag to skip padding
Encrypted RTP buffers may contain encrypted padding, hence it's
necessary to have an option to relax the validation in order to
successfully map the buffer.

When the flag GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING is set
gst_rtp_buffer_map() will map the buffer like if padding is not
present.

https://bugzilla.gnome.org/show_bug.cgi?id=752705
2015-10-15 22:42:37 +03:00
Sebastian Dröge
79d4765525 Revert "rtpbuffer: increase logging level when map fails"
This reverts commit e3c8a82017.

It causes too much noise in the logs.
2015-10-15 22:40:50 +03:00
Miguel París Díaz
e3c8a82017 rtpbuffer: increase logging level when map fails
https://bugzilla.gnome.org/show_bug.cgi?id=756641
2015-10-15 16:48:08 +03:00
Sebastian Dröge
b60ab758e4 Revert "audioencoder: timestamp headers same as first buffer and use duration 0"
This reverts commit dd4d6d9ed5.

It breaks ogg muxing and the vorbisenc unit test.
2015-10-12 14:02:58 +03:00
Havard Graff
dd4d6d9ed5 audioencoder: timestamp headers same as first buffer and use duration 0
https://bugzilla.gnome.org/show_bug.cgi?id=754224
2015-10-11 11:04:53 +01:00
Sebastian Dröge
4115814f36 Update GLib dependency to 2.40.0 2015-10-02 22:19:52 +03:00
Sebastian Rasmussen
042e71a117 rtpbasepayload: Implement video SDP attributes
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726472
2015-10-02 17:44:14 +03:00
Vineeth TM
8e5f7f27f5 gstreamer: base: Fix memory leaks when context parse fails.
When g_option_context_parse fails, context and error variables are not getting free'd
which results in memory leaks. Free'ing the same.

And replacing g_error_free with g_clear_error, which checks if the error being passed
is not NULL and sets the variable to NULL on free'ing.

https://bugzilla.gnome.org/show_bug.cgi?id=753852
2015-10-02 17:26:27 +03:00
Luis de Bethencourt
e90464d2f9 visual: make private all variable subclasses don't need
Subclasses don't need access to all variables. Making them private.

https://bugzilla.gnome.org/show_bug.cgi?id=742875
2015-10-01 16:05:58 +01:00
Luis de Bethencourt
8ae0fd3990 visual: merge audiovisalizer base classes
Move the audiovisualizer base class to pbutils, so it can be used by plugins
from other modules

https://bugzilla.gnome.org/show_bug.cgi?id=742875
2015-10-01 16:05:58 +01:00
Perry Hung
c5c2466d36 app: pass PKG_CONFIG_PATH for gir files for libgstapp as well
gir include search directories should respect PKG_CONFIG_PATH,
just like we do everywhere else. Makes g-i pick up the right
paths when using ./configure --with-pkg-config-path=

https://bugzilla.gnome.org/show_bug.cgi?id=755494
2015-09-29 13:10:35 +01:00
Sebastian Dröge
e49c492a53 Release 1.6.0 2015-09-25 23:15:20 +02:00
Sebastian Dröge
8239da2311 video-dither: Use saturated add when adding ordered dither for > 8 bit targets
Otherwise our 16 bit integers are going to overflow in intermediate
calculations, causing video to become mostly black.

https://bugzilla.gnome.org/show_bug.cgi?id=755392
2015-09-24 23:01:58 +02:00
Sebastian Dröge
1fb85733f5 video-frame: Fix gst_video_frame_copy() for formats with pstride==0
v210, UYVP and IYU1 are complex formats for which pixel stride does not really
have a meaning. If we copy width*pstride bytes per line, it's not going to do
the right thing. As a fallback, copy stride bytes per line. This might copy
uninitialized bytes at the end of each line, but at least copies the frame.

https://bugzilla.gnome.org/show_bug.cgi?id=755392
2015-09-24 17:39:43 +02:00
Aurélien Zanelli
94e0279c44 fdmemory: remove 'allow-none' annotation in gst_fd_allocator_alloc() doc
gst_fd_allocator_alloc() ensure that passed allocator is a fd memory
allocator, so that we can't pass NULL allocator.

https://bugzilla.gnome.org/show_bug.cgi?id=754833
2015-09-23 10:01:41 -04:00
Aurélien Zanelli
4559feb69e dmabuf: remove 'allow-none' annotation in gst_dmabuf_allocator_alloc() doc
gst_dmabuf_allocator_alloc() ensure that passed allocator is a DMABuf
allocator, so that we can't pass NULL allocator.

https://bugzilla.gnome.org/show_bug.cgi?id=754833
2015-09-23 10:01:41 -04:00
Christophe Fergeau
f096f1f9d3 appsink: Fix 'steaming' typo in API doc
There are several occurrences of 'steaming' where 'streaming' was meant.
2015-09-18 17:48:49 +02:00
Vineeth T M
060f0c21f2 audiosink, multisocketsink: Fix error leak during failures
https://bugzilla.gnome.org/show_bug.cgi?id=755143
2015-09-17 11:59:35 +02:00
Tim-Philipp Müller
a7bcdc3608 appsink: minor docs fix 2015-09-15 16:32:45 +01:00
Tim-Philipp Müller
8b96b52a62 videopool: ensure allocation alignment is consistent with video alignment requirements
Make sure GstAllocationParams alignment is not less than
any alignment requirement specified via GstVideoAlignment.

https://bugzilla.gnome.org/show_bug.cgi?id=754120
2015-09-15 10:20:37 +01:00
Thiago Santos
df931fb0b8 video-format: add missing alpha flag for some formats
Some formats didn't have the alpha flag marked, use the correct
macro so they get it right.

https://bugzilla.gnome.org/show_bug.cgi?id=754808
2015-09-10 07:28:21 -03:00