Vincent Penquerc'h [Mon, 28 Nov 2011 13:26:53 +0000 (13:26 +0000)]
Revert "libgstvideo: add a new API to handle QoS events and dropping logic"
This reverts commit
eb03323fb683e06ed8e7f557037f13252f150c25.
*grumble* I managed to merge something I did not mean to.
Vincent Penquerc'h [Mon, 28 Nov 2011 12:51:22 +0000 (12:51 +0000)]
various: fix pad template leaks
https://bugzilla.gnome.org/show_bug.cgi?id=662664
Vincent Penquerc'h [Wed, 7 Sep 2011 15:04:14 +0000 (16:04 +0100)]
theoradec: move the QoS logic to libgstvideo
https://bugzilla.gnome.org/show_bug.cgi?id=658241
Vincent Penquerc'h [Mon, 5 Sep 2011 12:56:05 +0000 (13:56 +0100)]
libgstvideo: add a new API to handle QoS events and dropping logic
https://bugzilla.gnome.org/show_bug.cgi?id=658241
Mark Nauwelaerts [Mon, 28 Nov 2011 10:30:18 +0000 (11:30 +0100)]
audioencoder: elaborate some documentation
Mark Nauwelaerts [Mon, 28 Nov 2011 10:28:06 +0000 (11:28 +0100)]
audiodecoder: add some documentation
Mark Nauwelaerts [Mon, 21 Nov 2011 13:26:54 +0000 (14:26 +0100)]
audiodecoder: really discard NULL decoded frame altogether
... including any timestamp, rather than having that one influence base_ts.
Stefan Sauer [Mon, 28 Nov 2011 09:55:39 +0000 (10:55 +0100)]
alsasrc: style fix
Use timestamp==0 instead of mixing it with !timestamp style checks.
Stefan Sauer [Mon, 28 Nov 2011 08:12:37 +0000 (09:12 +0100)]
alsasrc: handle the case where the drivers don't supply timestamps
If highres-timestamp is 0, try lowres and if that fails fallback to system clock
timestamps.
Vincent Penquerc'h [Tue, 1 Nov 2011 15:21:54 +0000 (15:21 +0000)]
oggmux: set collectpads2 not to wait on sparse streams
https://bugzilla.gnome.org/show_bug.cgi?id=663174
Josep Torra [Fri, 25 Nov 2011 14:35:39 +0000 (15:35 +0100)]
playsinkconvertbin: make identiy silent
Tim-Philipp Müller [Fri, 25 Nov 2011 13:11:54 +0000 (13:11 +0000)]
audio: remove unstable API guards from the audio decoder and encoder base classes
Tim-Philipp Müller [Fri, 25 Nov 2011 12:58:22 +0000 (12:58 +0000)]
docs: mention explicitly that playbin2 signals are emitted from a streaming thread
Sebastian Dröge [Fri, 25 Nov 2011 10:11:12 +0000 (11:11 +0100)]
decodebin2: Set the multiqueue limits to the playing limits after overrun too
We don't expect any new pads anymore and prerolling is finished now.
Sebastian Dröge [Fri, 25 Nov 2011 10:08:58 +0000 (11:08 +0100)]
decodebin2: Cache the upstream seekability for demuxer decode chains and use it for the non-preroll multiqueue limits
After preroll the multiqueue limits are still set to the preroll
limits if use-buffering is set to TRUE. In that case we only want
time limits on the multiqueue if upstream is seekable.
Vincent Penquerc'h [Tue, 8 Nov 2011 13:55:58 +0000 (13:55 +0000)]
decodebin2: fix prerolling for low bitrate streams from hlsdemux
Such streams were detected as seekable, as the query on the typefind
element was testing the m3u8 file listing the actual streams, and
not going through the demuxer(s).
We now check for seekability for each multiqueue following a demuxer,
so the query will flow through the elements which might prevent seeking.
https://bugzilla.gnome.org/show_bug.cgi?id=647769
Vincent Penquerc'h [Mon, 24 Oct 2011 10:46:05 +0000 (11:46 +0100)]
oggdemux: minor cleanup
Vincent Penquerc'h [Tue, 27 Sep 2011 15:45:26 +0000 (16:45 +0100)]
libgstriff: add a couple tags that need skipping
Found in a sample in the wild, appears to be ID3 tag.
https://bugzilla.gnome.org/show_bug.cgi?id=660249
Sebastian Dröge [Thu, 24 Nov 2011 13:41:13 +0000 (14:41 +0100)]
videorate: Rename ARG_ enums to PROP_
This is more consistent with other code and these are
properties anyway, not arguments
Sebastian Dröge [Thu, 24 Nov 2011 13:29:49 +0000 (14:29 +0100)]
videorate: Add property to force an output framerate
API: GstVideoRate:force-fps
Changing the framerate during playback is not possible
with a capsfilter downstream if upstream is not using
gst_pad_alloc_buffer(). In that case there's no way in
0.10 to signal to videorate that the preferred framerate
has changed.
This new property will force the output framerate to
a specific value and can be changed during playback.
Sebastian Dröge [Thu, 24 Nov 2011 11:38:54 +0000 (12:38 +0100)]
playsinkconvertbin: Reconfigure if we switch from raw to incompatible raw caps
We might need to add converters and worked in passthrough mode before.
Sebastian Dröge [Thu, 24 Nov 2011 11:37:58 +0000 (12:37 +0100)]
playsinkconvertbin: Override acceptcaps function for the two ghostpads
The ghostpad acceptcaps functions are not valid in this case because
we don't only accept the caps accepted by the target but could also
insert converters. Fixes bug #663892.
Sebastian Dröge [Thu, 24 Nov 2011 10:34:12 +0000 (11:34 +0100)]
playsinkaudioconvert: use-volume and use-converters are no construct-only properties anymore
Fixes bug #663893.
Vincent Penquerc'h [Sat, 22 Oct 2011 19:29:26 +0000 (20:29 +0100)]
oggdemux: skip the second bisection when possible
If we already saw the keyframes that we need to find,
we do not need to bisect to find them.
This will always be the case for streams with audio only,
where each frame acts as a keyframe, but will occasionally
also happen for streams with video.
https://bugzilla.gnome.org/show_bug.cgi?id=662475
Vincent Penquerc'h [Sat, 22 Oct 2011 19:20:38 +0000 (20:20 +0100)]
oggdemux: improve push time seeking
Various tweaks to improve convergence, in particular for
the worst case, which is now cut in about half.
https://bugzilla.gnome.org/show_bug.cgi?id=662475
Vincent Penquerc'h [Fri, 21 Oct 2011 18:38:19 +0000 (19:38 +0100)]
oggdemux: gather some more stats about bisection
https://bugzilla.gnome.org/show_bug.cgi?id=662475
Vincent Penquerc'h [Wed, 23 Nov 2011 16:09:13 +0000 (16:09 +0000)]
vorbisenc: do not accept 256 channels, 255 is the max vorbis supports
Vincent Penquerc'h [Tue, 22 Nov 2011 13:29:10 +0000 (13:29 +0000)]
oggstream: extract opus comments if available
Vincent Penquerc'h [Tue, 22 Nov 2011 13:15:33 +0000 (13:15 +0000)]
oggstream: recognize opus headers from data, not packet count
Opus streams outside of Ogg may not have headers, and oggstream
may be used by oggmux to mux an Opus stream which does not come
from Ogg - thus without headers.
Determining headerness by packet count would strip the first two
packets from such an Opus stream, leading to a very small amount
of audio being clipped at the beginning of the stream.
Vincent Penquerc'h [Tue, 22 Nov 2011 13:01:35 +0000 (13:01 +0000)]
oggdemux: add some more debug info when determining start time
Vincent Penquerc'h [Tue, 22 Nov 2011 12:55:56 +0000 (12:55 +0000)]
oggstream: fix opus duration calculation
Vincent Penquerc'h [Tue, 22 Nov 2011 12:00:58 +0000 (12:00 +0000)]
oggstream: early out on headers when determining packet duration
Vincent Penquerc'h [Mon, 21 Nov 2011 17:03:21 +0000 (17:03 +0000)]
oggstream: account for opus pre-skip in granpos/time mapping
René Stadler [Tue, 22 Nov 2011 09:04:12 +0000 (10:04 +0100)]
playsinkconvertbin: avoid removing children from bin twice
GstBin base class removes children in dispose, so we need to do the same.
Vincent Penquerc'h [Sat, 19 Nov 2011 16:06:09 +0000 (16:06 +0000)]
ogg: add opus support
Mark Nauwelaerts [Wed, 16 Nov 2011 18:00:44 +0000 (19:00 +0100)]
vorbisenc: reset tag setter interface when appropriate
Mark Nauwelaerts [Wed, 16 Nov 2011 18:00:30 +0000 (19:00 +0100)]
audioencoder: invalidate format info when setup negotiation failed
... which ensures nothing subsequently tries to slip past _chain
and into a possibly improperly setup subclass.
Vincent Penquerc'h [Tue, 15 Nov 2011 13:29:31 +0000 (13:29 +0000)]
audiodecoder: accept dropped buffers before we know the format
This allows flacdec to not emit audio for headers, while allowing
the base audio decoder to keep its timestamps in sync.
Robert Swain [Mon, 14 Nov 2011 11:45:31 +0000 (12:45 +0100)]
audio: Remove some unused variables
Olivier Crête [Tue, 30 Aug 2011 22:27:09 +0000 (18:27 -0400)]
rtcpbuffer: Add feedback message types from RFC 5104
These are Codec Control messages (CCM)
https://bugzilla.gnome.org/show_bug.cgi?id=658419
Mark Nauwelaerts [Wed, 19 Oct 2011 14:30:27 +0000 (16:30 +0200)]
audiodecoder: improve reverse playback
... by doing some more (reverse) timestamp interpolating and
refactoring downstream pushing.
Fixes #661983.
Tim-Philipp Müller [Sun, 13 Nov 2011 13:18:16 +0000 (13:18 +0000)]
audio: add GST_AUDIO_INFO_IS_VALID macro and use in audio decoder base class
API: GST_AUDIO_INFO_IS_VALID
Tim-Philipp Müller [Sat, 12 Nov 2011 15:51:52 +0000 (15:51 +0000)]
tests: require Gtk+ 3.0 for examples and Gtk-based test apps
The Gtk+ dependency is entirely optional, we're just not
supporting Gtk+ 2.x any longer.
Tim-Philipp Müller [Mon, 7 Nov 2011 17:36:44 +0000 (17:36 +0000)]
audio: fix order in LIBADD
Local libs must come first.
Tim-Philipp Müller [Fri, 11 Nov 2011 13:32:23 +0000 (13:32 +0000)]
playsinkconvertbin: fix visualisations again
Make caps writable before merging other caps into them.
Vincent Penquerc'h [Thu, 10 Nov 2011 15:55:31 +0000 (15:55 +0000)]
baseaudiosink: make unsigned properties unsigned, not signed
Tim-Philipp Müller [Wed, 9 Nov 2011 00:36:51 +0000 (00:36 +0000)]
configure: suppress warnings about unused variables if debugging system is disabled in core
https://bugzilla.gnome.org/show_bug.cgi?id=662952
Vincent Penquerc'h [Thu, 27 Oct 2011 13:48:52 +0000 (14:48 +0100)]
textoverlay: continue processing text when silent
This prevents playback wegding when text buffers are
left to pile up.
https://bugzilla.gnome.org/show_bug.cgi?id=662829
Tim-Philipp Müller [Tue, 8 Nov 2011 00:16:56 +0000 (00:16 +0000)]
win32: update .def file for new audiosink API
API: gst_base_audio_sink_get_alignment_threshold()
API: gst_base_audio_sink_set_alignment_threshold()
API: gst_base_audio_sink_get_discont_wait()
API: gst_base_audio_sink_set_discont_wait()
Tim-Philipp Müller [Mon, 7 Nov 2011 23:41:33 +0000 (23:41 +0000)]
examples: sprinkle GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS in seek test utility
https://bugzilla.gnome.org/show_bug.cgi?id=630497
Tim-Philipp Müller [Mon, 7 Nov 2011 23:05:44 +0000 (23:05 +0000)]
docs: fix up some Since: markers
Vincent Penquerc'h [Fri, 4 Nov 2011 10:34:27 +0000 (10:34 +0000)]
theoraenc: fix speed level failure test
It was testing the opposite of what it thought it was.
https://bugzilla.gnome.org/show_bug.cgi?id=663390
Vincent Penquerc'h [Fri, 4 Nov 2011 10:57:40 +0000 (10:57 +0000)]
theoraenc: make logically static const data just so
https://bugzilla.gnome.org/show_bug.cgi?id=663391
Vincent Penquerc'h [Fri, 4 Nov 2011 10:58:15 +0000 (10:58 +0000)]
theoraenc: use th_packet_iskeyframe instead of peeking at bits
https://bugzilla.gnome.org/show_bug.cgi?id=663391
Vincent Penquerc'h [Fri, 4 Nov 2011 10:59:00 +0000 (10:59 +0000)]
theoraenc: trivial comment typos fixes
https://bugzilla.gnome.org/show_bug.cgi?id=663391
Vincent Penquerc'h [Fri, 4 Nov 2011 10:59:12 +0000 (10:59 +0000)]
theoraenc: warn when trying to set an ignored obsolete property
https://bugzilla.gnome.org/show_bug.cgi?id=663391
Vincent Penquerc'h [Fri, 4 Nov 2011 11:10:46 +0000 (11:10 +0000)]
theoraenc: refuse to get to READY if the encoder was disabled
https://bugzilla.gnome.org/show_bug.cgi?id=663391
Vincent Penquerc'h [Tue, 18 Oct 2011 16:58:49 +0000 (17:58 +0100)]
oggdemux: survive skeleton finding length behind our backs in push mode
In push mode, we determine duration by doing a seek to the end of the
stream. However, a skeleton stream with an index will cause the duration
to be known already, and we end up never setting the push_time_duration
variable which we use to know duration has been determined.
https://bugzilla.gnome.org/show_bug.cgi?id=662049
Vincent Penquerc'h [Wed, 5 Oct 2011 14:29:54 +0000 (15:29 +0100)]
valgrind: add ALSA leaks fixed by snd_config_update_free_global
If they go when calling snd_config_update_free_global, they're
not really bug leaks, but more like intentional ones we don't
want to get told about.
https://bugzilla.gnome.org/show_bug.cgi?id=615342
Felipe Contreras [Mon, 2 May 2011 10:05:28 +0000 (13:05 +0300)]
baseaudiosink: make discont-wait configurable
Now we can configure how much time to wait before deciding that a
discont has happened.
Also, adds getter and setter to allow derived implementations to set
this value upon construction.
Suggestions and several improvements by Havard Graff.
Signed-off-by: Felipe Contreras <felipe.contreras@gmail.com>
Felipe Contreras [Mon, 7 Nov 2011 10:31:47 +0000 (11:31 +0100)]
baseaudiosink: delay the resyncing of timestamp vs ringbuffertime
A common problem for audio-playback is that the timestamps might not
be completely linear. This is specially common when doing streaming over
a network, where you can have jittery and/or bursty packettransmission,
which again will often be reflected on the buffertimestamps.
Now, the current implementation have a threshold that says how far the
buffertimestamp is allowed o drift from the ideal aligned time in the
ringbuffer. This was an instant reaction, and ment that if one buffer
arrived with a timestamp that would breach the drift-tolerance, a resync
would take place, and the result would be an audible gap for the
listener.
The annoying thing would be that in the case of a "timestamp-outlier",
you would first resync one way, say +100ms, and then, if the next
timestamp was "back on track", you would end up resyncing the other way
(-100ms) So in fact, when you had only one buffer with slightly off
timestamping, you would end up with *two* audible gaps. This is the
problem this patch addresses.
The way to "fix" this problem with the previous implementation, would
have been to increase the "drift-tolerance" to a value that was greater
than the largest timestamp-outlier one would normally expect. The big
problem with this approach, however, is that it will allow normal
operations with a huge offset timestamp vs running-time, which is
detrimental to lip-sync. If the drift-tolerance is set to 200ms, it
basically means that lip-sync can easily end up being off by that much.
This patch will basically start a timer when the first breach of
drift-tolerance is detected. If any following timestamp for the next n
nanoseconds gets "back on track" within the threshold, it has basically
eliminated the effect of an outlier, and the timer is stopped. If,
however, all timestamps within this time-limit are breaching the
threshold, we are probably facing a more permanent offset in the
timestamps, and a resync is allowed to happen.
So basically this patch offers something as rare as both higher
accuracy, it terms of allowing smaller drift-tolerances, as well as much
smoother, less glitchy playback!
Commit message and improvments by Havard Graff.
Fixes bug #640859.
Felipe Contreras [Mon, 7 Nov 2011 10:18:34 +0000 (11:18 +0100)]
baseaudiosink: rename some variables
Felipe Contreras [Sat, 21 May 2011 13:16:42 +0000 (16:16 +0300)]
baseaudiosink: use gst_util_uint64_scale_int when appropriate
It's probably safer this way.
Felipe Contreras [Sat, 21 May 2011 12:49:20 +0000 (15:49 +0300)]
baseaudiosink: split drift-tolerance into alignment-threshold
So that drift-tolerance is used for clock slaving resync, and
alignment-threshold is for timestamp drift.
Felipe Contreras [Sat, 21 May 2011 13:02:36 +0000 (16:02 +0300)]
baseaudiosink: trivial comment fixes
Some found by Havard Graff.
Signed-off-by: Felipe Contreras <felipe.contreras@gmail.com>
Sebastian Dröge [Fri, 4 Nov 2011 09:37:12 +0000 (10:37 +0100)]
subtitleoverlay: Use gst_caps_merge() instead of gst_caps_union()
This keeps the caps order and is more efficient.
Sebastian Dröge [Fri, 4 Nov 2011 09:36:51 +0000 (10:36 +0100)]
playsinkconvertbin: Use gst_caps_merge() instead of gst_caps_union()
This keeps the caps order and is more efficient.
Reynaldo H. Verdejo Pinochet [Fri, 4 Nov 2011 00:35:38 +0000 (21:35 -0300)]
Add missing default include paths to androgenizer call
Fixes building tag/ with Android's NDK
Mart Raudsepp [Thu, 3 Nov 2011 12:10:31 +0000 (14:10 +0200)]
decodebin2: Post all source pads in stream-topology messages as "element-srcpad" values
This allows us to easily get ahold of all pads on a stream-topology message, including
pre-decoder ones, while "pad" only gives us access to the raw pads (as used by discoverer).
Mart Raudsepp [Thu, 20 Oct 2011 10:04:52 +0000 (13:04 +0300)]
decodebin2: Use existing "caps" quark for one of the structure sets
Sebastian Dröge [Thu, 3 Nov 2011 09:07:27 +0000 (10:07 +0100)]
playsinkconvertbin: Don't add identity multiple times
Vincent Penquerc'h [Wed, 19 Oct 2011 13:13:39 +0000 (14:13 +0100)]
playsink: send flush start/stop event when we switch elements
https://bugzilla.gnome.org/show_bug.cgi?id=661262
Vincent Penquerc'h [Wed, 19 Oct 2011 13:13:30 +0000 (14:13 +0100)]
playsink: re-add identity where appropriate
https://bugzilla.gnome.org/show_bug.cgi?id=661262
Vincent Penquerc'h [Wed, 19 Oct 2011 13:12:01 +0000 (14:12 +0100)]
playsink: lock the new {set,get}_property functions
https://bugzilla.gnome.org/show_bug.cgi?id=661262
Thiago Santos [Mon, 17 Oct 2011 23:14:54 +0000 (23:14 +0000)]
playsinkconvertbin: Be more consistent with ghostpad targets
Set up targets on READY->PAUSED state change to passthrough by
default. This prevents the targets from being unset on the
first run, while the 'raw' variable would mean that some
target is set.
Thiago Santos [Mon, 17 Oct 2011 22:41:49 +0000 (22:41 +0000)]
playsinkconvertbin: No need to remove the identity
The identity element should be handled by the GstBin's cleanup,
removing it on the remove_elements function might remove it
too soon, as this function can be called directly from playsink
Thiago Santos [Mon, 17 Oct 2011 22:41:11 +0000 (22:41 +0000)]
playsinkconvertbin: Adding some debug messages
Adds a couple debug messages and some g_assert to make debugging
easier
Thiago Santos [Mon, 17 Oct 2011 22:02:03 +0000 (22:02 +0000)]
playsink-videoconvert: Fix warning on build
Remove unused variable
Vincent Penquerc'h [Mon, 17 Oct 2011 21:05:30 +0000 (21:05 +0000)]
playsink: handle after-the-fact changes in converters/volume booleans
The playsink was nastily poking a boolean in the structure.
Make those booleans properties, so we are told when they change,
and rebuild the conversion bin when they do.
Some cleanup to go with it too.
https://bugzilla.gnome.org/show_bug.cgi?id=661262
Vincent Penquerc'h [Mon, 17 Oct 2011 18:43:06 +0000 (18:43 +0000)]
playsink: handle NULL cached caps in getcaps
https://bugzilla.gnome.org/show_bug.cgi?id=661262
Vincent Penquerc'h [Mon, 17 Oct 2011 18:06:00 +0000 (18:06 +0000)]
playsink: consider both passthrough and converter caps in getcaps
Since we can switch between both modes.
https://bugzilla.gnome.org/show_bug.cgi?id=661262
Vincent Penquerc'h [Mon, 17 Oct 2011 17:54:27 +0000 (17:54 +0000)]
playsink: cache inner converter bin caps
https://bugzilla.gnome.org/show_bug.cgi?id=661262
Vincent Penquerc'h [Mon, 17 Oct 2011 17:26:48 +0000 (17:26 +0000)]
playsink: keep both raw and non raw pipelines at all times
and switch between them as needed.
https://bugzilla.gnome.org/show_bug.cgi?id=661262
Vincent Penquerc'h [Mon, 17 Oct 2011 17:29:50 +0000 (17:29 +0000)]
playsink: only compare against the media type we expect
ie, audio/x-raw- for audio, video/x-raw- for video.
Add a trailing - to be more specific. I doubt there's anything
like audio/x-rawhide or something, but you never know.
https://bugzilla.gnome.org/show_bug.cgi?id=661262
Vincent Penquerc'h [Mon, 17 Oct 2011 16:55:30 +0000 (16:55 +0000)]
playsink: refactor the converter bins since they are almost identical
https://bugzilla.gnome.org/show_bug.cgi?id=661262
Vincent Penquerc'h [Mon, 17 Oct 2011 13:00:05 +0000 (13:00 +0000)]
playsink: fix passthrough mode (hopefully)
The code was doing counterintuitive rewiring of pads when the
bin did not contain any elements. We now add an identity element
in that case, which makes it simpler, and should fix the AC3
passthrough mode when using pulseaudio (but I don't see the bug
here so can't test).
https://bugzilla.gnome.org/show_bug.cgi?id=661262
Vincent Penquerc'h [Fri, 7 Oct 2011 11:16:44 +0000 (11:16 +0000)]
playsink: handle NULL ghost pad target
For the src pad anyway.
https://bugzilla.gnome.org/show_bug.cgi?id=661262
Sebastian Dröge [Thu, 3 Nov 2011 08:56:14 +0000 (09:56 +0100)]
Revert "playsinkaudioconvert: Fix warning when there is no target pad yet"
This reverts commit
f35c51c14915729f0fdf2b348f351ea7e81027cc.
Better patch coming soon.
Sebastian Dröge [Fri, 28 Oct 2011 08:07:42 +0000 (10:07 +0200)]
oggmux: Remove obsolete #include
Tim-Philipp Müller [Wed, 2 Nov 2011 23:33:18 +0000 (23:33 +0000)]
docs: add draft for subtitle overlays to design docs
Main purpose is to provide a generic way to make subtitles work on
top of non-raw video (vaapi, vdpau, etc.).
Colin Walters [Wed, 2 Nov 2011 19:31:11 +0000 (15:31 -0400)]
configure: Allow setting GLIB_EXTRA_CFLAGS
Similar to gstreamer commit
bb2020b1e794210cf7d44c6626122f611016a620
Tim-Philipp Müller [Sun, 30 Oct 2011 20:00:47 +0000 (20:00 +0000)]
subtitleoverlay: don't use soon-to-be-deprecated gst_filter_run()
Mersad Jelacic [Fri, 28 Oct 2011 11:58:47 +0000 (13:58 +0200)]
audiosink: avoid deadlocking audioringbuffer thread
... when it goes into wait for ringbuffer starting just after such
having been signalled.
Fixes #661738.
Philip Jägenstedt [Tue, 26 Apr 2011 20:20:29 +0000 (22:20 +0200)]
typefind: extract SOF marker in jpeg typefinder
The SOF types are defined by http://www.w3.org/Graphics/JPEG/itu-t81.pdf
This is needed to make sure that we plug a jpeg decoder that
can handle the type of JPEG we have (e.g. lossless JPEG)
https://bugzilla.gnome.org/show_bug.cgi?id=556648
Thiago Santos [Mon, 10 Aug 2009 01:48:29 +0000 (01:48 +0000)]
oggmux: port to gstcollectpads2
Jan Schmidt [Thu, 27 Oct 2011 12:39:31 +0000 (23:39 +1100)]
build: Fix build for moved volume subdir
Stefan Sauer [Thu, 27 Oct 2011 07:51:46 +0000 (09:51 +0200)]
volume: move volume example to audio
Stefan Sauer [Thu, 27 Oct 2011 07:42:36 +0000 (09:42 +0200)]
audio examples. fix the makefile
Stefan Sauer [Thu, 27 Oct 2011 07:33:55 +0000 (09:33 +0200)]
volume: make global vars static
Stefan Sauer [Thu, 27 Oct 2011 07:33:01 +0000 (09:33 +0200)]
audiomix: add a simple audiomix example