Thibault Saunier [Fri, 18 Nov 2011 16:22:52 +0000 (13:22 -0300)]
textoverlay: Make the text_image data a buffer
This way we won't free data that would be attached to some buffer.
Thibault Saunier [Fri, 18 Nov 2011 14:04:47 +0000 (11:04 -0300)]
textoverlay: Sync the caps with the new supported formats
Thanks to the use of the new video composition library, we gain support to
more colospaces and formats, let's state it.
Thibault Saunier [Wed, 16 Nov 2011 20:54:43 +0000 (17:54 -0300)]
textoverlay: Make use of the new video blending utility
Tim-Philipp Müller [Fri, 25 Nov 2011 16:46:09 +0000 (16:46 +0000)]
tests: add basic unit test for video overlay composition and rectangles
Tim-Philipp Müller [Sat, 12 Nov 2011 14:59:35 +0000 (14:59 +0000)]
video: add video overlay composition API for subtitles
Basic API to attach overlay rectangles to buffers,
or blend them directly onto raw video buffers.
To be used primarily for things like subtitles or
logo overlays, not meant to replace videomixer.
Allows us to associate subtitle overlays with
non-raw video surface buffers, so that subtitles
are not lost and can instead be rendered later
when those surfaces are displayed or converted,
whilst re-using all the existing overlay plugins
and not having to teach them about our special
video surfaces. Could also have been made part
of the surface buffer abstraction of course, but
a secondary goal was to consolidate the blending
code for raw video into libgstvideo, and this
kind of API allows us to do both in a way that's
minimally invasive to existing elements, and at
the same time is fairly intuitive.
More features and extensions like the ability to
pass the source data or text/markup directly will
be added later.
https://bugzilla.gnome.org/show_bug.cgi?id=665080
API: gst_video_buffer_get_overlay_composition()
API: gst_video_buffer_set_overlay_composition()
API: gst_video_overlay_composition_new()
API: gst_video_overlay_composition_add_rectangle()
API: gst_video_overlay_composition_n_rectangles()
API: gst_video_overlay_composition_get_rectangle()
API: gst_video_overlay_composition_make_writable()
API: gst_video_overlay_composition_copy()
API: gst_video_overlay_composition_ref()
API: gst_video_overlay_composition_unref()
API: gst_video_overlay_composition_blend()
API: gst_video_overlay_rectangle_new_argb()
API: gst_video_overlay_rectangle_get_pixels_argb()
API: gst_video_overlay_rectangle_get_pixels_unscaled_argb()
API: gst_video_overlay_rectangle_get_render_rectangle()
API: gst_video_overlay_rectangle_set_render_rectangle()
API: gst_video_overlay_rectangle_copy()
API: gst_video_overlay_rectangle_ref()
API: gst_video_overlay_rectangle_unref()
Tim-Philipp Müller [Wed, 23 Nov 2011 00:31:18 +0000 (00:31 +0000)]
video: hide private video-blend.[ch] from gobject-introspection
And remove unused fields from helper structure.
Tim-Philipp Müller [Tue, 15 Nov 2011 18:00:00 +0000 (18:00 +0000)]
video: add fallbacks for compilation without orc
Thibault Saunier [Mon, 17 Oct 2011 15:25:11 +0000 (17:25 +0200)]
video: add some internal helper functions for image blending
This could be improved if we decide we don't need it to
be this generic/flexible.
Sebastian Dröge [Mon, 5 Dec 2011 08:38:33 +0000 (09:38 +0100)]
xoverlay: Fix mistakes in the sample code
Fixes bug #665430.
Tim-Philipp Müller [Sun, 4 Dec 2011 20:50:25 +0000 (20:50 +0000)]
Suppress deprecation warnings in selected files, for g_static_rec_mutex_* mostly
GStaticRecMutex is part of our API/ABI, not much we can do here
in 0.10 for most of these.
Tim-Philipp Müller [Sun, 4 Dec 2011 20:38:19 +0000 (20:38 +0000)]
alsamixer: use GRectMutext instead of GStaticRecMutex with newer glib versions
Tim-Philipp Müller [Sun, 4 Dec 2011 20:21:26 +0000 (20:21 +0000)]
alsamixer: embed static mutexes into the mixer structure
instead of allocating them dynamically
Tim-Philipp Müller [Sun, 4 Dec 2011 17:02:39 +0000 (17:02 +0000)]
tools, tests: g_thread_init() is deprecated in glib master
It's not needed any longer.
Tim-Philipp Müller [Sun, 4 Dec 2011 16:43:38 +0000 (16:43 +0000)]
Work around deprecated thread API in glib master
Add private replacements for deprecated functions such as
g_mutex_new(), g_mutex_free(), g_cond_new() etc., mostly
to avoid the deprecation warnings. We'll change these
over to the new API once we depend on glib >= 2.32.
Replace g_thread_create() with g_thread_try_new().
Tim-Philipp Müller [Sun, 4 Dec 2011 15:23:21 +0000 (15:23 +0000)]
xmpwriter: update for thread API deprecations in glib master
Stefan Sauer [Sun, 4 Dec 2011 12:43:06 +0000 (13:43 +0100)]
fft-example: re-add Makefile.am
Stefan Sauer [Fri, 2 Dec 2011 22:35:50 +0000 (23:35 +0100)]
configure: trim trailing whitespace
Stefan Sauer [Fri, 2 Dec 2011 22:34:47 +0000 (23:34 +0100)]
tests: add a test for fft result value-ranges
Add a small example that uses ffts of various types and parameters and check the
result value ranges.
Piotr Fusik [Tue, 13 Sep 2011 19:10:43 +0000 (21:10 +0200)]
various: typo fixes
Fix typos in code and docs. Fixes. #658984
Stefan Sauer [Thu, 1 Dec 2011 10:59:17 +0000 (11:59 +0100)]
adder: be more graceful in the clipfunction
Doing dynamic pipelines is hard in 0.10. As we don't have the sticky events in
0.10 and sending such events in special elements like adder and tee was outvoted
on last attempt, be graceful to the misbehaviour instead.
Tim-Philipp Müller [Thu, 1 Dec 2011 01:22:19 +0000 (01:22 +0000)]
tests: fix caps leak in audioresample tests
Tim-Philipp Müller [Thu, 1 Dec 2011 01:07:26 +0000 (01:07 +0000)]
tests: fix memory leak in basetime test
Tim-Philipp Müller [Wed, 30 Nov 2011 23:58:19 +0000 (23:58 +0000)]
playbin2: tone down debug message about file URIs with spaces
Complain a bit less loudly about URIs that have not been
escaped properly.
Tim-Philipp Müller [Wed, 30 Nov 2011 23:15:35 +0000 (23:15 +0000)]
Revert "alsasrc: Improve timestamp accuracy"
This reverts commit
0b774e0b7cf7a8ef1780fb6100228ca6e8ca8bcf.
Tim-Philipp Müller [Wed, 30 Nov 2011 23:15:22 +0000 (23:15 +0000)]
Revert "alsasrc: Fix some compilation errors"
This reverts commit
2b84f5bd74ddb50f7832917ea8b4dd38d005631b.
Tim-Philipp Müller [Wed, 30 Nov 2011 23:15:12 +0000 (23:15 +0000)]
Revert "alsa: Remove unused but set variable"
This reverts commit
e9aed7f31c7e9e415f733e147140ce3ef2f57a61.
Tim-Philipp Müller [Wed, 30 Nov 2011 23:15:03 +0000 (23:15 +0000)]
Revert "alsasrc: fail gracefully when ALSA does not give timestamps"
This reverts commit
c7282a5718c7f31f84fb31b2c38fab0f9a38e2b0.
Tim-Philipp Müller [Wed, 30 Nov 2011 23:14:54 +0000 (23:14 +0000)]
Revert "alsasrc: handle the case where the drivers don't supply timestamps"
This reverts commit
8154b69112cdc4830cd6002ec6c1f2917d30437b.
Stefan Sauer [Mon, 28 Nov 2011 09:55:39 +0000 (10:55 +0100)]
Revert "alsasrc: style fix"
This reverts commit
f70ca6d4cbfd2b672dcc7215814bf6b39ce2c3f8.
Sebastian Dröge [Wed, 30 Nov 2011 13:25:11 +0000 (14:25 +0100)]
playsinkconvertbin: Don't send undefined NEWSEGMENT events to the internal elements
This happens when the internal elements are added before any NEWSEGMENT
event arrived and in that case we shouldn't send a NEWSEGMENT event
to the internal elements at all. They will get the NEWSEGMENT event
from upstream later.
Sebastian Dröge [Tue, 29 Nov 2011 13:15:45 +0000 (14:15 +0100)]
playbin2: Fix decoder-sink compatibility check for raw audio/video formats
If the sink supports raw audio/video, we first check
if the decoder could output any raw audio/video format
and assume it is compatible with the sink then. We don't
do a complete compatibility check here if converters
are plugged between the decoder and the sink because
the converters will convert between raw formats and
even if the decoder format is not supported by the decoder
a converter will convert it.
We assume here that the converters can convert between
any raw format.
Fixes bug #665120.
Alessandro Decina [Tue, 29 Nov 2011 08:11:21 +0000 (09:11 +0100)]
oggdemux: fix compiler warning
Alessandro Decina [Tue, 29 Nov 2011 07:49:53 +0000 (08:49 +0100)]
libgstvideo: minor fixes to key unit events
Make out args to gst_video_event_parse_{downstream|upstream}_force_key_unit
optional, update libgstvideo.def and fix docs a bit.
API: gst_video_event_new_upstream_force_key_unit
API: gst_video_event_new_downstream_force_key_unit
API: gst_video_event_is_force_key_unit
API: gst_video_event_parse_upstream_force_key_unit
API: gst_video_event_parse_downstream_force_key_unit
https://bugzilla.gnome.org/show_bug.cgi?id=607742
Andoni Morales Alastruey [Sat, 4 Jun 2011 23:49:38 +0000 (01:49 +0200)]
libgstvideo: Add force key unit events
Philippe Normand [Mon, 28 Nov 2011 19:11:09 +0000 (20:11 +0100)]
fft: Bracket public headers
This is especially needed if the gstfftw library is used from C++
code.
Fixes #665074
Philippe Normand [Mon, 28 Nov 2011 19:10:18 +0000 (20:10 +0100)]
typefindfunctions: Fix compiler warning
Alexey Fisher [Mon, 28 Nov 2011 18:03:50 +0000 (19:03 +0100)]
typefind: fix build error
fix build errors:
gsttypefindfunctions.c:248:25: error: 'low' may be used uninitialized in this function [-Werror=uninitialized]
gsttypefindfunctions.c:239:24: error: 'high' may be used uninitialized in this function [-Werror=uninitialized]
Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
Sebastian Dröge [Mon, 28 Nov 2011 18:06:57 +0000 (19:06 +0100)]
playsinkconvertbin: Fix stupid mistake in last commit
Sebastian Dröge [Mon, 28 Nov 2011 18:03:54 +0000 (19:03 +0100)]
playsinkconvertbin: Only return the converter caps if we actually have raw caps
Fixes bug #664818 (hopefully).
Kipp Cannon [Mon, 28 Nov 2011 16:59:32 +0000 (17:59 +0100)]
audioresample: Don't emit DISCONT buffers if no discontinuity happened
audioresample is derived from GstBaseTransform, and one of
GstBaseTransform's traits is that if the derived element does not
produce an output buffer from some input buffer then the first output
buffer after that gets flaged as a discontinuity, whether or not the
buffer actually is discontinuous from the output buffer that preceded
it. When downsampling, the audioresample element requires more than
one input sample for each output sample, and if the ratio of input to
output sample rates is high enough and the input buffers short enough
it can come to pass that the resampler does not receive enough samples
on its input to produce any output. Currently the resampler returns
GST_BASE_TRANSFORM_FLOW_DROPPED from the transform() method in this case,
causing the next buffer to be flagged as a discontinuity. If subsequent
elements in the pipeline reset themselves on disconts, this can cause
clicks and other undesireable behaviour.
Fixes bug #665004.
Vincent Penquerc'h [Fri, 30 Sep 2011 19:00:50 +0000 (20:00 +0100)]
typefind: typefind UTF-16 and UTF-32
This avoids the MP3 typefinder from getting the highest score
every time it thinks there's something it might possibly be
able to parse.
https://bugzilla.gnome.org/show_bug.cgi?id=607619
Vincent Penquerc'h [Mon, 28 Nov 2011 13:27:29 +0000 (13:27 +0000)]
Revert "theoradec: move the QoS logic to libgstvideo"
This reverts commit
149a4ce390a78e21309b210f7daba9db5d42afe6.
*grumble* I managed to merge something I did not mean to.
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