Thibault Saunier [Thu, 5 Nov 2015 10:34:07 +0000 (11:34 +0100)]
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
Wim Taymans [Thu, 5 Nov 2015 09:40:18 +0000 (10:40 +0100)]
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()
Andreas Frisch [Mon, 10 Nov 2014 10:11:37 +0000 (11:11 +0100)]
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
Arnaud Vrac [Sat, 30 Nov 2013 00:59:55 +0000 (01:59 +0100)]
video: blend using OVER operation
Also support all premultiplied/non-premultiplied source/destination
configurations
https://bugzilla.gnome.org/show_bug.cgi?id=681447
Sebastian Dröge [Tue, 3 Nov 2015 14:51:47 +0000 (16:51 +0200)]
oggdemux: Create full Opus caps with all fields
https://bugzilla.gnome.org/show_bug.cgi?id=757152
Sebastian Dröge [Tue, 3 Nov 2015 16:30:09 +0000 (18:30 +0200)]
codec-utils: Add utilities for Opus caps and the OpusHead header
https://bugzilla.gnome.org/show_bug.cgi?id=757152
Sebastian Dröge [Tue, 3 Nov 2015 09:11:57 +0000 (11:11 +0200)]
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
Sebastian Dröge [Tue, 3 Nov 2015 08:58:35 +0000 (10:58 +0200)]
oggdemux: Add GstAudioClippingMeta for Opus for accurate start/end clipping
https://bugzilla.gnome.org/show_bug.cgi?id=757153
Sebastian Dröge [Mon, 2 Nov 2015 14:19:42 +0000 (16:19 +0200)]
audio: Add GstAudioClippingMeta for specifying clipping on encoded audio buffers
https://bugzilla.gnome.org/show_bug.cgi?id=757153
Sebastian Dröge [Mon, 2 Nov 2015 09:19:23 +0000 (11:19 +0200)]
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
Tim-Philipp Müller [Tue, 3 Nov 2015 16:38:09 +0000 (16:38 +0000)]
audio: update disted orc backup files
Luis de Bethencourt [Tue, 3 Nov 2015 14:08:25 +0000 (14:08 +0000)]
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
Luis de Bethencourt [Tue, 3 Nov 2015 13:44:39 +0000 (13:44 +0000)]
videodecoder: Print GstClockTimeDiff as a signed integer in debug logs
Wim Taymans [Tue, 3 Nov 2015 10:59:09 +0000 (11:59 +0100)]
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.
Wim Taymans [Tue, 3 Nov 2015 10:57:32 +0000 (11:57 +0100)]
audiopack: improve pack functions
Avoid shifts by using convh functions.
Wim Taymans [Tue, 3 Nov 2015 10:44:54 +0000 (11:44 +0100)]
audioconvert: change multiplier for int<->float conversion
Use (1 << 31) as the multiplier for int<->float conversions. This makes
sure that int->float conversions always end up with floats between
[-1.0, 1.0].
For the conversion from float to int, this multiplier will give the complete
int range after we perform clipping.
Change the unit test to take this into consideration.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755301
Luis de Bethencourt [Mon, 2 Nov 2015 17:32:55 +0000 (17:32 +0000)]
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
Luis de Bethencourt [Mon, 2 Nov 2015 16:36:35 +0000 (16:36 +0000)]
oggmux: Print GstClockTimeDiff as a signed integer in debug logs
Luis de Bethencourt [Mon, 2 Nov 2015 16:09:52 +0000 (16:09 +0000)]
oggdemux: Use GstClockTimeDiff and print signed integer in debug logs
Use GstClockTimeDiff and Clock macros to print signed integer time
differences in the debug logs.
https://bugzilla.gnome.org/show_bug.cgi?id=757480
Luis de Bethencourt [Mon, 2 Nov 2015 14:06:39 +0000 (14:06 +0000)]
examples: 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
Sebastian Dröge [Mon, 2 Nov 2015 15:14:51 +0000 (17:14 +0200)]
audio: Fix parameters to gst_buffer_get_audio_downmix_meta() in macro
Wim Taymans [Mon, 2 Nov 2015 14:54:19 +0000 (15:54 +0100)]
audiotestsrc: increase freq limit
Raise the frequency limit and try to negotiate to a samplerate of 4*freq
when larger then the default samplerate.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=754450
Wim Taymans [Mon, 2 Nov 2015 14:46:22 +0000 (15:46 +0100)]
audiotestsrc: add support for unlimited number of channels
Raise the channel limit and set the channel-mask for > 2 channels.
Wim Taymans [Mon, 2 Nov 2015 12:19:09 +0000 (13:19 +0100)]
audiotestsrc: add support for all formats
Use the pack functions to also support the other audio formats we
have.
Luis de Bethencourt [Mon, 2 Nov 2015 12:09:42 +0000 (12:09 +0000)]
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
Thibault Saunier [Mon, 2 Nov 2015 10:34:56 +0000 (11:34 +0100)]
encoding-profile: Do not force user to provide an encoding profile name
And use the profile called `default` if none provided.
Thibault Saunier [Mon, 2 Nov 2015 10:30:07 +0000 (11:30 +0100)]
encoding-target: Do not unconditionally break when searching for a target
Otherwise the loop is useless!
Fixes CID 1338051
Sebastian Dröge [Sat, 24 Oct 2015 17:08:47 +0000 (20:08 +0300)]
audioresample: Clip input buffers to the segment before handling them
https://bugzilla.gnome.org/show_bug.cgi?id=757068
Sebastian Dröge [Sat, 24 Oct 2015 17:05:10 +0000 (20:05 +0300)]
audioconvert: Clip input buffers to the segment before handling them
https://bugzilla.gnome.org/show_bug.cgi?id=757068
Sebastian Dröge [Sat, 24 Oct 2015 17:02:13 +0000 (20:02 +0300)]
audiofilter: Clip input buffers to the segment before handling them
https://bugzilla.gnome.org/show_bug.cgi?id=757068
Tim-Philipp Müller [Sun, 1 Nov 2015 23:05:10 +0000 (23:05 +0000)]
audioconvert: update orc backup code to fix build without orc
Csaba Toth [Mon, 26 Oct 2015 20:32:41 +0000 (21:32 +0100)]
multisocketsink: fix "client-removed" signal on 64-bit platforms and with bindings
The client-removed signal used G_INT_TYPE instead of G_SOCKET_TYPE
in its definition leading to problems on platforms where the size
of a pointer is larger than the size of an integer, It would also
not work at all with dynamic language bindings.
https://bugzilla.gnome.org/show_bug.cgi?id=757155
Joan Pau Beltran [Wed, 28 Oct 2015 17:36:41 +0000 (18:36 +0100)]
videotestsrc: fix handling of Bayer format 'gbrg'
Due to a typo, videotestsrc did not handle the Bayer
format 'gbrg' properly and reported it as invalid,
causing negotiation errors.
https://bugzilla.gnome.org/show_bug.cgi?id=757264
Wim Taymans [Fri, 30 Oct 2015 16:36:48 +0000 (17:36 +0100)]
audioconvert: rework audioconvert
Rewrite audioconvert to try to make it more clear what steps are
executed during conversion.
Add passthrough step that just does a memcpy when possible.
Add ORC optimized dither and quantization functions.
Implement noise-shaping on S32 samples only and allow for arbitrary
noise shaping coefficients if we want this later.
Wim Taymans [Fri, 30 Oct 2015 16:33:32 +0000 (17:33 +0100)]
channelmix: fix up API a little
don't use gpointer * for something that should be gpointer.
Wim Taymans [Wed, 28 Oct 2015 10:40:42 +0000 (11:40 +0100)]
audioquantize: make helper for add with saturation
Sebastian Dröge [Thu, 29 Oct 2015 14:52:31 +0000 (16:52 +0200)]
videodecoder: Print another time difference as a signed integer instead of a huge unsigned one
Sebastian Dröge [Thu, 29 Oct 2015 14:01:26 +0000 (16:01 +0200)]
videodecoder: Print GstClockTimeDiff as a signed integer in debug logs
Nirbheek Chauhan [Wed, 28 Oct 2015 18:31:01 +0000 (00:01 +0530)]
tools: gst-device-monitor: fix two memory leaks
The removed GList link needs to be freed too, and
the G_OPTION_REMAINING arguments need to be freed.
Thibault Saunier [Wed, 28 Oct 2015 14:50:44 +0000 (15:50 +0100)]
encoding-target: Add a GST_ENCODING_TARGET_PATH envvar to find target files
Thibault Saunier [Wed, 28 Oct 2015 14:47:00 +0000 (15:47 +0100)]
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.
Thibault Saunier [Tue, 27 Oct 2015 11:56:48 +0000 (12:56 +0100)]
encoding-target: Create directory before trying to save encoding targets
Thibault Saunier [Tue, 27 Oct 2015 11:50:26 +0000 (12:50 +0100)]
encoding-profile: Allow specifying the target category in the serialized encoding target
Wim Taymans [Tue, 27 Oct 2015 16:28:06 +0000 (17:28 +0100)]
audioconvert: make the quantizer a reusable object
Turn the quantizer into a reusable object.
Wim Taymans [Tue, 27 Oct 2015 12:24:31 +0000 (13:24 +0100)]
audioconvert: make the channel mixer a separate reusable object
A first attempt at making the channel mixer a separate object.
Wim Taymans [Wed, 28 Oct 2015 10:32:57 +0000 (11:32 +0100)]
audioquantize: fix 8-pole noise shaping
Fix the 8-pole noise shaping error update. We were mixing errors from
different channels.
Sebastian Dröge [Tue, 27 Oct 2015 13:44:06 +0000 (15:44 +0200)]
decodebin: Send SEEK events directly to adaptive streaming demuxers
This makes sure that they will always get SEEK events, even if we're currently
in the middle of a group switch (i.e. switching to another
representation/bitrate/etc).
https://bugzilla.gnome.org/show_bug.cgi?id=606382
Guillaume Desmottes [Tue, 6 Oct 2015 13:20:51 +0000 (15:20 +0200)]
decodebin: fix event leak
As stated in GST_PAD_PROBE_HANDLED's documentation, we are
supposed to unref the event before returning.
Fixes an event leak in the validate.hls.playback.play_15s.hls_bibbop
validate scenario.
https://bugzilla.gnome.org/show_bug.cgi?id=754459
Sebastian Dröge [Fri, 23 Oct 2015 16:13:05 +0000 (19:13 +0300)]
audioconvert: Update disted orc files
Wim Taymans [Fri, 23 Oct 2015 14:58:17 +0000 (16:58 +0200)]
audioconvert: use pack/unpack functions
Rework the converter to use the pack/unpack functions
Because the unpack functions can only unpack to 1 format, add a separate
conversion step for doubles when the unpack function produces int.
Do conversion to S32 in the quantize function directly.
Tweak the conversion factor for doing float->int conversion slightly to
get the full range of negative samples, use clamp to make sure we don't
exceed our int range on the positive axis (see also #755301)
Sebastian Dröge [Fri, 23 Oct 2015 09:02:28 +0000 (12:02 +0300)]
playbin: Send upstream events directly to playsink
Send event directly to playsink instead of letting GstBin iterate
over all sink elements. The latter might send the event multiple times
in case the SEEK causes a reconfiguration of the pipeline, as can easily
happen with adaptive streaming demuxers.
What would then happen is that the iterator would be reset, we send the
event again, and on the second time it will fail in the majority of cases
because the pipeline is still being reconfigured
Eunhae Choi [Fri, 23 Oct 2015 08:25:50 +0000 (17:25 +0900)]
tests: typefindfunctions: fix error leaks
https://bugzilla.gnome.org/show_bug.cgi?id=757008
Thibault Saunier [Wed, 23 Sep 2015 16:47:52 +0000 (18:47 +0200)]
videotestsrc: Force alpha downstream if foreground color contains alpha
Otherwise the foreground color won't be fully represented in the
outputted frames.
https://bugzilla.gnome.org/show_bug.cgi?id=755482
Pavel Bludov [Thu, 22 Oct 2015 04:07:44 +0000 (12:07 +0800)]
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
Tim-Philipp Müller [Wed, 21 Oct 2015 13:34:56 +0000 (14:34 +0100)]
Automatic update of common submodule
From b99800a to b319909
Sebastian Dröge [Tue, 20 Oct 2015 14:29:42 +0000 (17:29 +0300)]
Use new GST_ENABLE_EXTRA_CHECKS #define
https://bugzilla.gnome.org/show_bug.cgi?id=756870
Sebastian Dröge [Wed, 21 Oct 2015 11:25:47 +0000 (14:25 +0300)]
Automatic update of common submodule
From 9aed1d7 to b99800a
Sebastian Dröge [Tue, 20 Oct 2015 09:08:23 +0000 (12:08 +0300)]
rtp: GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING is Since 1.6.1
Matthew Waters [Mon, 19 Oct 2015 16:58:26 +0000 (03:58 +1100)]
decodebin: track the exposable pads through connect_pad
The logic introduced by
[d50b713: decodebin: set the decode pad target before setting elements to PAUSED]
to expose pads would only ever be able to possibly expose one (the last) pad per element.
Make it so that any exposable pads are able to be exposed rather than just the
last pad returned by connect_element.
https://bugzilla.gnome.org/show_bug.cgi?id=742924
Matthew Waters [Mon, 19 Oct 2015 16:52:24 +0000 (03:52 +1100)]
decodebin: return the possibly new chain in analyze_new_pad
In the case of analyzing a demuxer chain, analyze_new_pad may create
a new GstDecodeChain. This was not propagated to the calling function which as
of [
d50b713f decodebin: set the decode pad target before setting elements to PAUSED]
is now required to be able to expose the correct pad.
https://bugzilla.gnome.org/show_bug.cgi?id=742924
Rajat Verma [Mon, 19 Oct 2015 10:02:19 +0000 (15:32 +0530)]
playsink: relink text_pad in case of reconfiguration
In case of reconfiguration, text_pad should be re-connected with
stream synchronizer sink pad. Otherwise we'll leave an unlinked pad around if
there always was a streamsynchronizer text pad.
https://bugzilla.gnome.org/show_bug.cgi?id=756804
eunhae choi [Mon, 14 Sep 2015 06:25:11 +0000 (15:25 +0900)]
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
Sebastian Dröge [Sat, 17 Oct 2015 19:25:22 +0000 (22:25 +0300)]
decodebin/playbin/playsink/subtitleoverlay: Post async-done on state change failures
https://bugzilla.gnome.org/show_bug.cgi?id=756611
Sebastian Dröge [Sat, 17 Oct 2015 19:20:31 +0000 (22:20 +0300)]
playsink: Immediately error out if state change fails
Otherwise we chain up to the parent class' change_state function and might
override the failure with SUCCESS.
https://bugzilla.gnome.org/show_bug.cgi?id=756611
Sebastian Dröge [Sat, 17 Oct 2015 18:47:07 +0000 (21:47 +0300)]
playbin/uridecodebin: Always post async-done immediately if we're a live pipeline
Not only if the base class told us, but also if one of our own elements did.
https://bugzilla.gnome.org/show_bug.cgi?id=756611
Matthew Waters [Thu, 15 Oct 2015 16:40:43 +0000 (03:40 +1100)]
decodebin: set the decode pad target before setting elements to PAUSED
Otherwise caps and context queries will disappear into nothing and therefore
fail. With autoplug-query now actually working, users (such as playbin) can
proxy these queries to the selected video sink and be able to select an
more appropriate configuration.
https://bugzilla.gnome.org/show_bug.cgi?id=731204
Sebastian Dröge [Sat, 17 Oct 2015 17:36:27 +0000 (20:36 +0300)]
video: Add out annotations to the out parameters of gst_video_calculate_display_ratio()
https://bugzilla.gnome.org/show_bug.cgi?id=754567
Matthew Waters [Thu, 15 Oct 2015 23:48:50 +0000 (10:48 +1100)]
win32 update exports for new rtp symbols
Stian Selnes [Wed, 22 Jul 2015 09:31:05 +0000 (11:31 +0200)]
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
Sebastian Dröge [Thu, 15 Oct 2015 19:40:50 +0000 (22:40 +0300)]
Revert "rtpbuffer: increase logging level when map fails"
This reverts commit
e3c8a820176ba39dfae85944fa9c6ae202ec681d.
It causes too much noise in the logs.
Miguel París Díaz [Thu, 15 Oct 2015 13:32:58 +0000 (15:32 +0200)]
rtpbuffer: increase logging level when map fails
https://bugzilla.gnome.org/show_bug.cgi?id=756641
Vineeth TM [Thu, 15 Oct 2015 01:01:38 +0000 (10:01 +0900)]
playsink: Fix volume element leak
In case sink implements a streamvolume interface, volume element is being got
from the sink. But this is transfer full. So the memory should be freed before
setting it to NULL. This was resulting in major memory leaks
https://bugzilla.gnome.org/show_bug.cgi?id=755867
Sebastian Dröge [Tue, 13 Oct 2015 21:32:11 +0000 (00:32 +0300)]
alsa: Use 8 bit pointer type for byte-based pointer arithmetic
Usually these loops only run once, so there's no problem here. But sometimes
they run twice, and by adding the number of bytes to a 16 bit pointer type we
would advance twice as much as we should.
Also use snd_pcm_frames_to_bytes() in alsasrc to calculate
the number of bytes to skip, same as we do in alsasink.
Thanks to Lucio A. Hernandez <lucio.a.hernandez@gmail.com> for reporting.
Sebastian Dröge [Mon, 12 Oct 2015 11:02:58 +0000 (14:02 +0300)]
Revert "audioencoder: timestamp headers same as first buffer and use duration 0"
This reverts commit
dd4d6d9ed54c2a63a7e45661519d9965417707c5.
It breaks ogg muxing and the vorbisenc unit test.
Havard Graff [Fri, 28 Aug 2015 09:44:19 +0000 (11:44 +0200)]
audioencoder: timestamp headers same as first buffer and use duration 0
https://bugzilla.gnome.org/show_bug.cgi?id=754224
Havard Graff [Fri, 28 Aug 2015 09:25:22 +0000 (11:25 +0200)]
audioencoder-tests: port to use GstHarness
https://bugzilla.gnome.org/show_bug.cgi?id=754223
Havard Graff [Thu, 27 Aug 2015 15:28:30 +0000 (17:28 +0200)]
audiodecoder-test: port to using GstHarness
https://bugzilla.gnome.org/show_bug.cgi?id=754196
Sebastian Dröge [Sun, 4 Oct 2015 17:36:00 +0000 (18:36 +0100)]
xvimagesink: Put error message into debug output instead of just throwing it away
Sebastian Dröge [Fri, 2 Oct 2015 19:19:52 +0000 (22:19 +0300)]
Update GLib dependency to 2.40.0
Sebastian Rasmussen [Sat, 15 Mar 2014 16:35:56 +0000 (17:35 +0100)]
rtpbasepayload: Implement video SDP attributes
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726472
Vivia Nikolaidou [Fri, 25 Sep 2015 12:17:53 +0000 (15:17 +0300)]
gst-play: Removed erroneous comment
The "fall through" comment was wrong. Removed.
https://bugzilla.gnome.org/show_bug.cgi?id=755440
Vivia Nikolaidou [Tue, 22 Sep 2015 20:12:10 +0000 (23:12 +0300)]
gst-play: Add keyboard shortcut '0' to seek to beginning
https://bugzilla.gnome.org/show_bug.cgi?id=755440
Vineeth T M [Tue, 25 Aug 2015 07:24:12 +0000 (16:24 +0900)]
videorate: remove unnecessary break statement
Trivial patch to remove unncessary break statement used after
goto statement.
https://bugzilla.gnome.org/show_bug.cgi?id=754054
Vineeth TM [Thu, 20 Aug 2015 06:59:15 +0000 (15:59 +0900)]
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
Mathieu Duponchelle [Wed, 24 Jun 2015 21:55:35 +0000 (23:55 +0200)]
encodebin: Fix special case
Allows to run such a command line :
gst-launch-1.0 uridecodebin uri=file:///home/meh/Music/sthg.mp4 ! \
encodebin profile-string="audio/x-wav|1" ! filesink location=sthg.wav
Previously the code failed because wavenc is considered as a muxer.
We still want encodebin to audio/x-wav as an AudioEncodingProfile,
so this simple fix allows that.
Ability to mux raw streams in containers such as matroskamux
is a different issue.
https://bugzilla.gnome.org/show_bug.cgi?id=751470
Rajat Verma [Tue, 29 Sep 2015 04:42:28 +0000 (10:12 +0530)]
decodebin: free hidden groups at time of switching groups
hidden groups should be freed at time of switching groups to avoid memory use
from balloning up.
https://bugzilla.gnome.org/show_bug.cgi?id=755770
Sebastian Dröge [Fri, 2 Oct 2015 07:07:33 +0000 (10:07 +0300)]
win32: Update exports for new audiovisualizer symbols
Jan Schmidt [Fri, 2 Oct 2015 05:04:34 +0000 (15:04 +1000)]
tests: Add baseaudiovisualizer test, moved from -bad
Jan Schmidt [Fri, 2 Oct 2015 05:05:26 +0000 (15:05 +1000)]
videotestsrc: Don't fixate framerate if downstream didn't provide one
intersection with a downstream that accepts any video/x-raw caps
with no further detail won't create a framerate field. If it's
not in the caps, don't fixate it, just set it to 30/1
Stefan Sauer [Thu, 1 Oct 2015 19:53:20 +0000 (21:53 +0200)]
docs: add alsamidisrc to docs
Antonio Ospite [Thu, 1 Oct 2015 19:43:21 +0000 (21:43 +0200)]
midi: add an ALSA MIDI sequencer source
The alsamidisrc element allows to get input event from ALSA MIDI
sequencer devices, and possibly convert them to sound using some
downstream element like fluiddec.
Fixes #738687
Luis de Bethencourt [Thu, 1 Oct 2015 14:27:55 +0000 (15:27 +0100)]
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
Luis de Bethencourt [Thu, 1 Oct 2015 10:55:59 +0000 (11:55 +0100)]
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
Tim-Philipp Müller [Thu, 1 Oct 2015 11:48:52 +0000 (12:48 +0100)]
typefinding: minor clean-up
Remove unnecessary brackets from IS_MPEGTS_HEADER macro.
Pankaj Darak [Thu, 1 Oct 2015 11:32:33 +0000 (12:32 +0100)]
typefinding: mpeg-ts detection improvement
Allow AFC to be 0 for null pid packets.
https://bugzilla.gnome.org/show_bug.cgi?id=726117
Tim-Philipp Müller [Wed, 30 Sep 2015 17:18:15 +0000 (18:18 +0100)]
tests: subparse: add unit test for closing tag detection
</ i> should be handled like </i>
https://bugzilla.gnome.org/show_bug.cgi?id=755875
Tim-Philipp Müller [Wed, 30 Sep 2015 17:17:13 +0000 (18:17 +0100)]
subparse: detect closing tags even if there's a space after the slash
</ i> should be handled like </i>
https://bugzilla.gnome.org/show_bug.cgi?id=755875
Perry Hung [Wed, 23 Sep 2015 15:59:22 +0000 (11:59 -0400)]
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
Sebastian Dröge [Fri, 25 Sep 2015 21:51:06 +0000 (23:51 +0200)]
Back to development
Sebastian Dröge [Fri, 25 Sep 2015 21:15:20 +0000 (23:15 +0200)]
Release 1.6.0