While this changes API slightly (e.g. actually uses set_format now), which is OK
for unstable API, it has following merits:
* symmetric w.r.t. stop at state change
* in line with other base class practice
* otherwise no subclass method at state change (global activation time)
Moreover, subclassese are either unaffected or trivially adjusted accordingly.
While this changes order w.r.t. set_format, which is OK for unstable API,
it has following merits:
* symmetric w.r.t. stop at state change
* in line with other base class practice
* little benefit in invoking 2 subclass virtual methods (set_format and start)
in immediate succession; all actions in the second could be done in the first
whereas subclass has no chance to do anything 'global' at activation time
Moreover, current -bad subclass relevant methods either trivially commute
or are either trivially adjusted accordingly.
Remove the android/ top dir
Fixe the Makefile.am to be androgenized
To build gstreamer for android we are now using androgenizer which generates the needed Android.mk files.
Androgenizer can be found here: http://git.collabora.co.uk/?p=user/derek/androgenizer.git
This is not implemented in any of our real sources to which wrappercamerabinsrc
might connect but this is optional and can be implemented at any time. A
limit on the software zoom level using video{crop,scale} would be arbitrary.
Use resource warning messages to notify camerabin2 that a capture
as aborted or couldn't be started, making it decrement the
processing counter and making the idle property more reliable.
Checks if the new received preview-caps are equal to what is
already in use, skips the preview-caps setting logic in case
new caps are same as current ones.
Adds a virtual function to basecamerasrc in case subclasses want to be
notified of changing preview caps. This is useful if the subclass wants
to post the preview itself or if it wants to provide a preview buffer
as close to as possible to the user's requested resolution to the
preview generation pipeline.
Adds some more logging and always assume capture has started before
start_capture is called. This helps on image captures that might
call finish_capture directly from start_capture or before start_capture
finishes.
Adds a readable property to gstphotography interface to query
what are the allowed preview caps supported.
Patch by Tommi Myöhänen <ext-tommi.1.myohanen@nokia.com>
The timestamps are only used if the output adapter is used, not
if complete frames are provided by the decoder and finish_frame() is
called and even in the case where the output adapter is used they
might not be used and are leaked.
Add some guards and fat warnings to the header files with still unstable
API, so people who just look at the installed headers know that it
actually is unstable API.
Merging previous commit into current codebase.
Move include directives for gst-libs into GST_PLUGINS_BAD_CFLAGS,
and fix all the Makefiles that use it. This is so that all the
include directories are added in the proper order: first the
directories in srcdir/builddir, then gst-plugins-base dirs, then
gstreamer dirs. If the order is wrong, installed headers may be
used instead of local headers and/or uninstalled headers from -base.
Because config.h defines __MSVCRT_VERSION__, which should be defined
before inclusion of any system header.
Also fixes mpegdemux Makefile.am LIBADD typo.
Fixes#606665
This allows to get rid of the sampling_rate variable in the base-class. Also now
subclasses can modify the caps to actualy negotiate. This is needed to e.g. set
audio-channel positions.
Revert the changes that added audio positions to template caps. We have an un-
fortunate limitation in core that does not allow to do it. Keep a few things
commented out, so that the channel position can later on be set in setcaps.
This reverts commit 4c087bcb07.
The reverted commit changes the order that set_format() and start()
are called, which is incorrect. The correct order is set_format(),
start(), handle_frame()..., stop()
This queries port roles from the LV2 data and converts it into GStreamer
channel positions. This should allow any type of multi-channel plugin
(including beyond stereo, e.g. surround) to work fine in GStreamer,
and with elements that require channel positions to be explicitly stated.
Install the headers, version the library with @GST_MAJORMINOR@,
add all required libraries to _LIBADD instead of _LDFLAGS,
and add GST_*_LDFLAGS to _LDFLAGS.
Fixes bug #594715.
-remove gst-libs/gst/dshow
-fakesource is moved from gst-libs/gst/dshow to sys/dshowsrcwrapper
-some minor changes (C/C++ check and includes) to make the plugin
compile again.
Add some guards and fat warnings to the header files with still unstable
API, so people who just look at the installed headers know that it
actually is unstable API.
Also move schroedinger plugin. This creates a new library,
gstbasevideo-0.10, which will probably be merged back into
gstvideo-0.10 when this is moved back to -base.
- Separate gstsignalprocessor into a separate library (not sure if this
is in the right place, but it works for now anyway)
- Create LV2 element based on LADSPA element, port most discovery
functionality
Application developers won't know right away which module an interface comes from,
and may assume that it is covered by the usual GStreamer API guarantees, so make
it as clear as possible that this particular API is still subject to change
(should have done that with other libraries in -bad before too really).
Original commit message from CVS:
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_init), (gst_app_src_set_property),
(gst_app_src_get_property), (gst_app_src_query),
(gst_app_src_set_latencies), (gst_app_src_set_latency),
(gst_app_src_get_latency), (gst_app_src_push_buffer_full):
* gst-libs/gst/app/gstappsrc.h:
Add properties and methods to configure and retrieve the min and max
latencies.
Original commit message from CVS:
* examples/app/appsrc-ra.c: (feed_data):
* examples/app/appsrc-seekable.c: (feed_data):
* examples/app/appsrc-stream.c: (read_data):
* examples/app/appsrc-stream2.c: (feed_data):
Fix example to unref after emiting the push-buffer action.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_push_buffer_full), (gst_app_src_push_buffer),
(gst_app_src_push_buffer_action):
Don't take the ref on the buffer in push-buffer action because it's too
awkward for bindings. Fixes#564482.
Original commit message from CVS:
Patch by: 이문형 <iwings at gmail dot com>
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_push_buffer):
Don't forget to release the lock again if we bail out because some
pad is flushing or we've reached EOS, otherwise things will lock up
next time _push_buffer() is called (#562802).
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c:
* gst/h264parse/gsth264parse.c:
Wim, you're a bad boy. You don't want people to contact you or what?
Original commit message from CVS:
2008-06-16 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
(gst_app_src_get_max_bytes, gst_app_src_push_buffer): Use
G_GUINT64_FORMAT. Avoid overflow in get_max_bytes().
Original commit message from CVS:
* examples/app/.cvsignore:
* examples/app/Makefile.am:
* examples/app/appsink-src.c: (on_new_buffer_from_source),
(on_source_message), (on_sink_message), (main):
Add beefed up example app from bug #413418. It now also uses appsink
instead of fakesink for more ultimate coolness.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_init), (gst_app_src_set_property),
(gst_app_src_get_property), (gst_app_src_unlock),
(gst_app_src_unlock_stop), (gst_app_src_create),
(gst_app_src_set_max_bytes), (gst_app_src_push_buffer),
(gst_app_src_end_of_stream):
* gst-libs/gst/app/gstappsrc.h:
Add block property to allow push based implementation to block when we
fill up the appsrc queues.
Emit the enough-data signal while releasing our lock.
Original commit message from CVS:
* ext/dc1394/gstdc1394.c:
* ext/ivorbis/vorbisdec.c:
* ext/jack/gstjackaudiosink.c:
* ext/metadata/gstmetadatademux.c:
* ext/mythtv/gstmythtvsrc.c:
* ext/theora/theoradec.c:
* gst-libs/gst/app/gstappsink.c:
* gst/bayer/gstbayer2rgb.c:
* gst/deinterlace/gstdeinterlace.c:
* gst/rawparse/gstaudioparse.c:
* gst/rawparse/gstvideoparse.c:
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/selector/gstinputselector.c:
* gst/selector/gstoutputselector.c:
* gst/videosignal/gstvideoanalyse.c:
* gst/videosignal/gstvideodetect.c:
* gst/videosignal/gstvideomark.c:
* sys/oss4/oss4-mixer.c:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-source.c:
Do not use short_description in section docs for elements. We extract
them from element details and there will be warnings if they differ.
Also fixing up the ChangeLog order.
Original commit message from CVS:
* examples/app/Makefile.am:
* examples/app/appsrc-ra.c: (feed_data), (seek_data),
(found_source), (bus_message), (main):
* examples/app/appsrc-seekable.c: (feed_data), (seek_data),
(found_source), (bus_message), (main):
* examples/app/appsrc-stream2.c: (feed_data), (found_source),
(bus_message), (main):
Added 3 more example application for using appsrc in random-access mode,
pull-mode streaming and pull mode seekable.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_start), (gst_app_src_do_get_size),
(gst_app_src_create):
* gst-libs/gst/app/gstappsrc.h:
Make stream-type property writable.
Unset flushing when starting so that we reuse appsrc.
Inform basesrc about the configured size.
Emit seek-data signal when we are going to a different offset in
random-access mode.
Original commit message from CVS:
* examples/app/.cvsignore:
* examples/app/Makefile.am:
* examples/app/appsrc-stream.c: (read_data), (start_feed),
(stop_feed), (found_source), (bus_message), (main):
Added an example on how to use appsrc in playbin in streaming mode from
an mmapped file.
* examples/app/appsrc_ex.c: (main):
Set pipeline to NULL to free queued buffers.
* gst-libs/gst/app/gstapp-marshal.list:
* gst-libs/gst/app/gstappsrc.c: (stream_type_get_type), (_do_init),
(gst_app_src_class_init), (gst_app_src_init),
(gst_app_src_flush_queued), (gst_app_src_dispose),
(gst_app_src_set_property), (gst_app_src_get_property),
(gst_app_src_unlock), (gst_app_src_unlock_stop),
(gst_app_src_start), (gst_app_src_stop), (gst_app_src_is_seekable),
(gst_app_src_check_get_range), (gst_app_src_do_seek),
(gst_app_src_create), (gst_app_src_set_stream_type),
(gst_app_src_get_stream_type), (gst_app_src_set_max_bytes),
(gst_app_src_get_max_bytes), (gst_app_src_push_buffer),
(gst_app_src_end_of_stream), (gst_app_src_uri_get_type),
(gst_app_src_uri_get_protocols), (gst_app_src_uri_get_uri),
(gst_app_src_uri_set_uri), (gst_app_src_uri_handler_init):
* gst-libs/gst/app/gstappsrc.h:
Measure max queue size in bytes instead.
Add support for 3 modes of operation, streaming, seekable and
random-access, making basesrc handle the scheduling modes for each.
Add appsrc:// uri handler so that automatic plugging can be done from
playbin2 or uridecodebin, for example.
Added support for custom segment formats.
Add support for push and pull based operations from the application.
Expand the methods so that errors can be detected.
Flush the queued buffers on seeks and when shutting down.
Add signals to inform the app that a seek must happen.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
(gst_app_sink_init), (gst_app_sink_set_property),
(gst_app_sink_get_property), (gst_app_sink_unlock_start),
(gst_app_sink_unlock_stop), (gst_app_sink_flush_unlocked),
(gst_app_sink_start), (gst_app_sink_stop), (gst_app_sink_event),
(gst_app_sink_preroll), (gst_app_sink_render),
(gst_app_sink_set_caps), (gst_app_sink_set_drop),
(gst_app_sink_get_drop):
* gst-libs/gst/app/gstappsink.h:
Start some docs.
Add property to drop buffers when the queue is filled
Fix unlocking and flushing when the queues are filled.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
(gst_app_sink_init), (gst_app_sink_set_property),
(gst_app_sink_get_property), (gst_app_sink_event),
(gst_app_sink_preroll), (gst_app_sink_render),
(gst_app_sink_set_emit_signals), (gst_app_sink_get_emit_signals),
(gst_app_sink_set_max_buffers), (gst_app_sink_get_max_buffers),
(gst_app_sink_pull_buffer):
* gst-libs/gst/app/gstappsink.h:
Add more docs.
Add signals for when preroll and render buffers are available.
Add property to control signal emission.
Add property to control the max queue size.
Original commit message from CVS:
* gst-libs/gst/dshow/Makefile.am:
Use CXXFLAGS rather than CFLAGS; these are C++ files.
Define required constants appropriately.
* sys/dshowdecwrapper/Makefile.am:
Add required include dir, libraries.
Define required constants appropriately.
Original commit message from CVS:
* gst-libs/gst/dshow/Makefile.am:
* sys/dshowdecwrapper/Makefile.am:
* sys/dshowsrcwrapper/Makefile.am:
Add Makefiles to win32 plugins and lib.
They will need to be tested and probably fixed by developers
working with mingw. This is a first step to include source files
with releases.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose):
Really clean up the queue instead of just unreffing all buffers
in it.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_base_init),
(gst_app_src_class_init), (gst_app_src_init),
(gst_app_src_dispose), (gst_app_src_finalize):
Fix dispose/finalize.
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* gst-libs/gst/dshow/gstdshowfakesink.cpp:
(CDshowFakeSink.CDshowFakeSink):
* gst-libs/gst/dshow/gstdshowfakesink.h: (CDshowFakeSink.m_hres):
Fix crasher in constructor due to the base class's constructor
not necessarily being NULL-safe (depends on the SDK version used
apparently; #492406).
* sys/dshowsrcwrapper/gstdshowaudiosrc.c: (gst_dshowaudiosrc_prepare):
* sys/dshowsrcwrapper/gstdshowvideosrc.c: (gst_dshowvideosrc_set_caps):
Fix a couple of MSVC compiler warnings (#492406).
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_marshal_OBJECT__VOID),
(gst_app_sink_class_init), (gst_app_sink_init),
(gst_app_sink_dispose), (gst_app_sink_finalize),
(gst_app_sink_set_property), (gst_app_sink_get_property),
(gst_app_sink_flush_unlocked), (gst_app_sink_start),
(gst_app_sink_event), (gst_app_sink_getcaps),
(gst_app_sink_set_caps), (gst_app_sink_get_caps),
(gst_app_sink_is_eos), (gst_app_sink_pull_preroll),
(gst_app_sink_pull_buffer):
* gst-libs/gst/app/gstappsink.h:
Add properties, signals and actions to access the element even without
linking to the library.
Fix some method names and signatures.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init):
Override the preroll vmethod instead of overriding the render method
twice.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_base_init),
(gst_app_sink_class_init), (gst_app_sink_dispose),
(gst_app_sink_flush_unlocked), (gst_app_sink_start),
(gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
(gst_app_sink_render), (gst_app_sink_get_caps),
(gst_app_sink_set_caps), (gst_app_sink_end_of_stream),
(gst_app_sink_pull_preroll), (gst_app_sink_pull_buffer):
* gst-libs/gst/app/gstappsink.h:
Make love to appsink.
Make it support pulling of the preroll buffer.
Add docs and debug statements.
Fix some races wrt to EOS handling and stopping.
Implement getcaps.
Implement FLUSHING.
API: gst_app_sink_pull_preroll()
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins.args:
Remove directsoundsink property doc as this sink use the mixer
interface now.
* docs/plugins/gst-plugins-bad-plugins.interfaces:
Add interfaces implemented by Windows sinks.
* sys/directsound/gstdirectsoundsink.c:
* sys/directsound/gstdirectsoundsink.h:
Remove directsoundsink property and implement the mixer interface.
* win32/vs6/gst_plugins_bad.dsw:
* win32/vs6/libgstdirectsound.dsp:
Update project files.
* gst-libs/gst/dshow/gstdshow.cpp:
* gst-libs/gst/dshow/gstdshow.h:
* gst-libs/gst/dshow/gstdshowfakesink.cpp:
* gst-libs/gst/dshow/gstdshowfakesink.h:
* gst-libs/gst/dshow/gstdshowfakesrc.cpp:
* gst-libs/gst/dshow/gstdshowfakesrc.h:
* gst-libs/gst/dshow/gstdshowinterface.cpp:
* gst-libs/gst/dshow/gstdshowinterface.h:
* win32/common/libgstdshow.def:
* win32/vs6/libgstdshow.dsp:
Add a new gst library which allow to create internal Direct Show
graph (pipelines) to wrap Windows sources, decoders or encoders.
It includes a DirectShow fake source and sink and utility functions.
* sys/dshowsrcwrapper/gstdshowaudiosrc.c:
* sys/dshowsrcwrapper/gstdshowaudiosrc.h:
* sys/dshowsrcwrapper/gstdshowsrcwrapper.c:
* sys/dshowsrcwrapper/gstdshowsrcwrapper.h:
* sys/dshowsrcwrapper/gstdshowvideosrc.c:
* sys/dshowsrcwrapper/gstdshowvideosrc.h:
* win32/vs6/libdshowsrcwrapper.dsp:
Add a new plugin to wrap DirectShow sources on Windows.
It gets data from any webcam, dv cam, micro. We could add
tv tunner card later.
Original commit message from CVS:
* gst-libs/gst/app/Makefile.am:
Use GST_ALL_LDFLAGS, which actually exists, but maybe David
can confirm that was what he wanted.
Original commit message from CVS:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/gstappbuffer.c:
* gst-libs/gst/app/gstappbuffer.h:
* gst-libs/gst/app/gstappsrc.c:
Add GstAppBuffer that includes a callback and closure for
proper handling of data chunks.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_caps_with_data):
Do actually fix invalid RIFF fmt header values for alaw
and mulaw audio instead of just saying so.
* gst/wavparse/gstwavparse.c: (gst_wavparse_fmt):
Give gst_riff_create_audio_caps_with_data() a chance to
fix up broken format header fields before extracting any
parameters from the header. (fixes#167633)
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data):
Add extradata to huffyuv (fixes#165013).
* gst-libs/gst/riff/riff-read.c:
(gst_riff_read_strf_vids_with_data):
Fix extradata extraction if it is in the chunk size.
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_peek_element_data),
(gst_riff_read_element_data):
* gst-libs/gst/riff/riff-read.h:
Add _peek version (req'ed in CDXA).
* gst/cdxaparse/gstcdxaparse.c: (gst_cdxaparse_init),
(gst_cdxaparse_loop):
Fix parsing in playbin.
* gst/playback/gstdecodebin.c: (close_pad_link):
Ignore current_ pads, they cause major annoyance.
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_use_event):
Don't bail on unknown events.
* gst/audioscale/gstaudioscale.c: (gst_audioscale_chain):
Don't crash on events before negotiation.
* gst/avi/gstavidemux.c: (gst_avi_demux_add_stream):
Send tags on pads, too.
* gst/playback/gststreamselector.c:
(gst_stream_selector_request_new_pad):
Forward events on first pad if no input was selected yet.
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
* gst/wavenc/riff.h:
Add AMR (VBR and CBR) ids to riff.h audio codec list
* gst/asfdemux/gstasfdemux.c:
(gst_asf_demux_process_ext_content_desc),
(gst_asf_demux_process_object):
Retrieve more tags from ASF files (Genre, AlbumTitle, Artist)
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst-libs/gst/resample/resample.c: (gst_resample_sinc_ft_s16):
Fix invalid memory access (#159211).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data):
Add BLZ0 (Blizzard's version of DivX) fourcc.
Original commit message from CVS:
* configure.ac: add audioresample and cairo plugins. Remove
HAVE_MMX stuff, because it's not used.
* ext/Makefile.am: same
* ext/audioresample/Makefile.am: You are not ready for an
audio resampling element based on audioresample.
* ext/audioresample/gstaudioresample.c:
* ext/audioresample/gstaudioresample.h:
* ext/cairo/Makefile.am: You are not ready for overlay elements
based on cairo. Don't look too closely, these elements kinda
suck right now.
* ext/cairo/gstcairo.c: new
* ext/cairo/gsttextoverlay.c: new
* ext/cairo/gsttextoverlay.h: new
* ext/cairo/gsttimeoverlay.c: new
* ext/cairo/gsttimeoverlay.h: new
* gst-libs/gst/media-info/media-info-priv.h: fix compile
problem with compilers that don't support variadic macros.
Original commit message from CVS:
* gst/asfdemux/README
* gst/wavenc/riff.h
* gst-libs/gst/riff/riff-ids.h
* gst-libs/gst/riff/riff-media.c
add new 4CC codes for h263 related codecs
fixes partially bug #155163
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_chanpos_from_gst),
(gst_faad_chanpos_to_gst), (gst_faad_chain):
Set DURATION even if source buffer didn't. Also use increasing
timestamps.
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_caps_with_data):
Block_align can have larger values than 8192.
Original commit message from CVS:
* ext/esd/esdsink.c: (gst_esdsink_chain):
Make error actually say something useful (fixes#156798).
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_video_template_caps):
Add Intel Video 5.0 fourcc (IV50).
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_use_event):
Don't forward DISCONT events (fixes#159684).
Original commit message from CVS:
2004-11-27 Martin Soto <martinsoto@users.sourceforge.net>
* gst-libs/gst/audio/audioclock.c (gst_audio_clock_set_active)
(gst_audio_clock_get_internal_time):
Fix active <-> inactive transitions: ensure time value always
grows and avoid abrupt value changes.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_get_caps_internal):
buffer-frames property was missing
* ext/arts/gst_arts.c:
rate missing from sinkcaps
* ext/audiofile/gstafparse.c:
* ext/audiofile/gstafsink.c:
* ext/audiofile/gstafsrc.c:
* ext/swfdec/gstswfdec.c:
int audio doesn't know buffer-frames
* ext/cdparanoia/gstcdparanoia.c:
int audio doesn't know chunksize either
* ext/nas/nassink.c:
it's endianness, not endianess
* gst-libs/gst/audio/audio.h:
make float standard pad template caps really describe float
* gst/law/mulaw.c: (linear_factory):
signed only, please
* gst/mpegstream/gstdvddemux.c:
widths of 20 are not valid
Original commit message from CVS:
* ext/dv/gstdvdec.c: (gst_dvdec_handle_sink_event):
Set EOS on the element when processing an EOS event.
* ext/speex/gstspeexdec.h:
* ext/speex/gstspeexenc.h:
Only keep a const ptr to the mode
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_caps_with_data),
(gst_riff_create_audio_template_caps):
Allow WMAV3, with up to 6 channels.
* gst/asfdemux/gstasfmux.c: (gst_asfmux_request_new_pad):
Don't call gst_pad_set_event_function on a sink pad.
* gst/mpegstream/gstdvddemux.c:
(gst_dvd_demux_get_subpicture_stream),
(gst_dvd_demux_set_cur_audio), (gst_dvd_demux_set_cur_subpicture):
Copy the explicit caps that were set across to the cur_* pads,
instead of trying to use a possibly non-existent negotiated caps.
Reset the type of subpicture pads to UNKNOWN after calling init_stream,
so that the caps get set.
Original commit message from CVS:
2004-10-28 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_link):
fix build
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_link):
fix link function to always query channels and query width for
floats
* configure.ac:
add equalizer dir
* gst/equalizer/Makefile.am:
* gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_get_type),
(gst_iir_equalizer_base_init), (gst_iir_equalizer_class_init),
(gst_iir_equalizer_init), (gst_iir_equalizer_finalize),
(arg_to_scale), (setup_filter),
(gst_iir_equalizer_compute_frequencies),
(gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property),
(gst_iir_equalizer_filter_inplace), (gst_iir_equalizer_setup),
(plugin_init):
add an equalizer
Original commit message from CVS:
* gst/wavparse/gstwavparse.c:
add ATRAC3 to STATIC CAPS to fix a warning
* gst/matroska/ebml-read.c:
* gst-libs/gst/riff/riff-read.c:
fix typos
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_video_template_caps):
Add wing commander format mimetype/fourccs.
* gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
Don't crash if some value is 0.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_video_template_caps):
Add DIB fourcc (raw, palettized 8-bit RGB).
* gst-libs/gst/riff/riff-read.c:
(gst_riff_read_strf_vids_with_data):
Oops, fix strf_data reading bug.
* gst/avi/gstavidemux.c: (gst_avi_demux_add_stream):
Use a non-NULL tag.
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
Time for hacks. Sorry Dave. At least one quicktime movie (a
trailer) that I've encountered contains multiple video tracks.
One of those is the actual video track, the other are one-frame
tracks (images). Unfortunately, the number of frames according
to the trak header is 1 for each, so that doesn't help. So
instead, I look at the duration and discard tracks with a
duration shorter than 20% of the length of the stream. Better
than nothing.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_audio_caps_with_data):
Add codec_data handling (like asfdemux used to do).
* gst/asfdemux/gstasf.c: (plugin_init):
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_base_init),
(gst_asf_demux_add_audio_stream), (gst_asf_demux_add_video_stream):
Use riff-media for caps creation instead of our own (mostly
broken) copy of its functions.
Original commit message from CVS:
* ext/dvdread/dvdreadsrc.c: (dvdreadsrc_class_init),
(dvdreadsrc_init), (dvdreadsrc_dispose), (dvdreadsrc_set_property),
(dvdreadsrc_get_property), (_open), (_seek), (_read),
(dvdreadsrc_get), (dvdreadsrc_open_file),
(dvdreadsrc_change_state):
Fix. Don't do one big huge loop around the whole DVD, that will
cache all data and thus eat sizeof(dvd) (several GB) before we
see something.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek):
Actually NULL'ify event after using it.
* gst/matroska/ebml-read.c: (gst_ebml_read_use_event),
(gst_ebml_read_handle_event), (gst_ebml_read_element_id),
(gst_ebml_read_element_length), (gst_ebml_read_element_data),
(gst_ebml_read_seek), (gst_ebml_read_skip):
Handle events.
* gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_base_init),
(gst_dvd_demux_init), (gst_dvd_demux_get_audio_stream),
(gst_dvd_demux_get_subpicture_stream), (gst_dvd_demux_plugin_init):
Fix timing (this will probably break if I seek using menus, but
I didn't get there yet). VOBs and normal DVDs should now work.
Add a mpeg2-only pad with high rank so this get autoplugged for
MPEG-2 movies.
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_base_init),
(gst_mpeg_demux_class_init), (gst_mpeg_demux_init),
(gst_mpeg_demux_new_output_pad), (gst_mpeg_demux_get_video_stream),
(gst_mpeg_demux_get_audio_stream),
(gst_mpeg_demux_get_private_stream), (gst_mpeg_demux_parse_packet),
(gst_mpeg_demux_parse_pes), (gst_mpeg_demux_plugin_init):
Use this as second rank for MPEG-1 and MPEG-2. Still use this for
MPEG-1 but use dvddemux for MPEG-2.
* gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_class_init),
(gst_mpeg_parse_init), (gst_mpeg_parse_new_pad),
(gst_mpeg_parse_parse_packhead):
Timing. Only add pad template if it exists. Add sink template from
class and not from ourselves. This means we will always use the
correct sink template even if it is not the one defined in this
file.
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flacdec_src_query):
Only return true if we actually filled something in. Prevents
player applications from showing a random length for flac files.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init),
(gst_riff_read_use_event), (gst_riff_read_handle_event),
(gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh),
(gst_riff_read_strf_vids_with_data),
(gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs):
OK, ok, so I implemented event handling. Apparently it's normal
that we receive random events at random points without asking
for it.
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
(gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index),
(gst_avi_demux_sync), (gst_avi_demux_stream_scan),
(gst_avi_demux_massage_index), (gst_avi_demux_stream_header),
(gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry),
(gst_avi_demux_stream_data), (gst_avi_demux_loop):
* gst/avi/gstavidemux.h:
Implement non-lineair chunk handling and subchunk processing.
The first solves playback of AVI files where the audio and video
data of individual buffers that we read are not synchronized.
This should not happen according to the wonderful AVI specs, but
of course it does happen in reality. It is also a prerequisite for
the second. Subchunk processing allows us to cut chunks in small
pieces and process each of these pieces separately. This is
required because I've seen several AVI files with incredibly large
audio chunks, even some files with only one audio chunk for the
whole file. This allows for proper playback including seeking.
This patch is supposed to fix all AVI A/V sync issues.
* gst/flx/gstflxdec.c: (gst_flxdec_class_init),
(flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop):
Work.
* gst/modplug/gstmodplug.cc:
Proper return value setting for the query() function.
* gst/playback/gstplaybasebin.c: (setup_source):
Being in non-playing state (after, e.g., EOS) is not necessarily
a bad thing. Allow for that. This fixes playback of short files.
They don't actually playback fully now, because the clock already
runs. This means that small files (<500kB) with a small length
(<2sec) will still not or barely play. Other files, such as mod
or flx, will work correctly, however.
Original commit message from CVS:
* ext/dirac/Makefile.am:
* ext/dirac/gstdirac.cc:
* ext/dirac/gstdiracdec.cc:
* ext/dirac/gstdiracdec.h:
Do something. Don't actually know if this works because I don't
have a demuxer yet.
* ext/gsm/gstgsmdec.c: (gst_gsmdec_getcaps):
Add channels=1 to caps returned from _getcaps().
* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_get_type),
(gst_ogm_video_parse_get_type), (gst_ogm_audio_parse_base_init),
(gst_ogm_video_parse_base_init), (gst_ogm_parse_init),
(gst_ogm_audio_parse_init), (gst_ogm_video_parse_init),
(gst_ogm_parse_sink_convert), (gst_ogm_parse_chain),
(gst_ogm_parse_change_state):
Separate between audio/video so ogmaudioparse actually uses the
audio pad templates. Both audio and video work now, including
autoplugging. Also use sometimes-srcpad hack.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek):
Handle events better. Don't hang on infinite loops.
* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
(gst_avi_demux_init), (gst_avi_demux_reset),
(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
(gst_avi_demux_stream_header), (gst_avi_demux_stream_data),
(gst_avi_demux_change_state):
* gst/avi/gstavidemux.h:
Improve A/V sync. Still not perfect.
* gst/matroska/ebml-read.c: (gst_ebml_read_seek),
(gst_ebml_read_skip):
Handle events better.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event),
(gst_qtdemux_loop_header), (qtdemux_parse_trak),
(qtdemux_audio_caps):
Add IMA4. Improve event handling. Save offset after a seek when
the headers are at the end of the file so that we don't end up in
an infinite loop.
* gst/typefind/gsttypefindfunctions.c: (qt_type_find):
Add low-priority typefind support for files with no length.
Original commit message from CVS:
2004-08-24 Sebastien Cote <sc5@hermes.usherb.ca>
* gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head),
(gst_riff_read_element_data), (gst_riff_read_seek),
(gst_riff_read_skip): fix infinite loop in wavparse, fixes bug
#144616, patch reviewed by Ronald and committed by Christophe Fergeau
<teuf@gnome.org>
Original commit message from CVS:
Remove GPL'ed mmx32idct.c code and supporting code, since logic in gst-plugins
is not supposed to be GPL'ed. This code provided MMX optimisations, but was
never compiled in since configure never set HAVE_LIBMMX anyway.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_caps_with_data):
Fix double end-to-native symbol conversion (#148021).
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c:
(gst_riff_read_strf_vids_with_data),
(gst_riff_read_strf_auds_with_data):
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
(gst_avi_demux_add_stream), (gst_avi_demux_stream_header):
Make sure we don't create 0 sized subbufers in riff-read.
Signal the no more pads signal after reading the avi header.
Original commit message from CVS:
* gst-libs/gst/video/video.h:
Added 32 bits RGBA. Not sure if we should use another mime-type
for alpha rgb. Currently the presence of the alpha_mask property
signals an alpha channel. Ronald?
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data):
mp42/mp43 (no caps) exist too.
* gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps):
Set pixel_width/height; we've got them in-caps.
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
* gst/wavparse/gstwavparse.c: (plugin_init):
Both are valid primary.
* sys/oss/gstossmixer.c:
Remove i18n hack and enable translations.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
(gst_riff_create_video_template_caps):
Fix the template caps to include some more media types.
Original commit message from CVS:
* gst-libs/gst/play/play.c: (gst_play_pipeline_setup),
(gst_play_get_length_callback), (gst_play_set_location),
(gst_play_seek_to_time), (gst_play_set_data_src),
(gst_play_set_video_sink), (gst_play_set_audio_sink),
(gst_play_set_visualization), (gst_play_connect_visualization),
(gst_play_get_sink_element):
- add debugging info
- fix looking up sink elements by iterating over complete caps
- put everything except for source and autoplugger in a complete bin
Original commit message from CVS:
* gst-libs/gst/colorbalance/Makefile.am:
* gst-libs/gst/mixer/Makefile.am:
* gst-libs/gst/play/Makefile.am:
* gst-libs/gst/tuner/Makefile.am:
* gst/tcp/Makefile.am:
* sys/dxr3/Makefile.am:
don't include -enumtypes.[ch] or -marshal.[ch] files in the disted
tarball.
Also add all *.list files that were missing.
* Makefile.am:
add a distcheck hook to ensure the above doesn't happen again.
Original commit message from CVS:
2004-06-15 Zaheer Abbas Merali <zaheerabbas at merali.org>
fixed a potential leak with previous commit
* gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head):
Original commit message from CVS:
2004-06-15 Zaheer Abbas Merali <zaheerabbas at merali.org>
* gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head):
Added missing refcount, fixes bug #144425
Cheers Tim for finding the bug
Original commit message from CVS:
* gst-libs/gst/tuner/tunerchannel.h:
- add a freq_multiplicator field to make the conversion
between internal frequency unit and Hz
* sys/v4l/gstv4lelement.c:
* sys/v4l2/gstv4l2element.c:
- change default video device to /dev/video0
* sys/v4l/v4l_calls.c:
* sys/v4l2/v4l2_calls.c:
- we only expose frequency to the user in Hz instead of
bastard v4lX unit (either 62.5kHz or 62.5Hz)
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
Initialise b_o_s and e_o_s variables
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data):
Add some unusual fourcc's from mplayer avi's
* gst/multipart/multipartmux.c: (gst_multipart_mux_plugin_init):
Make the muxer have rank GST_RANK_NONE, so it doesn't mess up
autoplugging.
Original commit message from CVS:
third batch :
remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc
(in gst-plugins/gst-libs/ this time)
Original commit message from CVS:
* gst-libs/gst/resample/resample.c: (gst_resample_sinc_ft_s16),
(gst_resample_sinc_ft_float): Remove use of static temporary
buffer. This code was obviously not supposed to last long, but
it's stuck in our ABI, so it required a little hack to make it
ABI-compatible. Fixes#142585.
* gst-libs/gst/resample/resample.h: same.
Original commit message from CVS:
reviewed by: Ronald Bultje <rbultje@ronald.bitfreak.net>
* gst-libs/gst/audio/audioclock.c:
Fix wrong return type (#142205).
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c :
- fix INFO tag extraction in RIFF/AVI files
because gst_event_unref (event) also freed taglist
- avoid a mem leak
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h :
- add CDXA to the list of RIFF types
- add plst (playlist ?) to wav chunk list (only diff with wavparse/riff.h)
Original commit message from CVS:
* gst/auparse/gstauparse.c:
fixes a-law, adds mu-law, linear pcm (8,16,24,32), ieee (32, 64)
only unsupported formats are ADPCM/CCITT G.72x
reviewed by Ronald
* gst-libs/gst/audio/audio.h:
adds 24bit depth to PCM (x-raw-int)
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_audio_caps),
(gst_riff_create_audio_template_caps):
* gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head):
Fix for unaligned RIFF files (i.e. where all the chunks together
in a LIST chunk are not of the same size as the size given in
the LIST chunk header). Fixes several odd WAVE files. Also fix
ADPCM (block_align property) in audio, so that wavparse based
on this works now as it used to stand-alone.
Original commit message from CVS:
Rewrote wavparse to use riff-read instead of doing bytestream stuff by hand.
Made some useful functions in riff-read non-static.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
audio/x-raw-int with height rules! not. Now it's depth.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data):
mpegversion is an int
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_base_init):
don't try to create pad templates with NULL caps, use any caps
instead.
Original commit message from CVS:
* gconf/Makefile.am: Fix for non-GNU make
* gst-libs/gst/Makefile.am: Change directory order to handle
GstPlay linking with gstinterfaces
* gst-libs/gst/audio/make_filter: make use of tr portable
* gst-libs/gst/play/Makefile.am: Add intended \
* gst-libs/gst/xwindowlistener/xwindowlistener.c:
(gst_xwin_set_clips): Switch to ISO variadic macro. Use a
function prototype instead of void *.
* gst/ffmpegcolorspace/gstffmpegcodecmap.c: Switch to ISO variadic
macro.
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcolorspace_chain): wrap NULL in GST_ELEMENT_ERROR call
* gst/videofilter/make_filter: make use of tr portable
* pkgconfig/Makefile.am: Remove GNU extension in Makefile target
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data):
Add div[3456] as fourccs for DivX 3 (fixes#140137).
Original commit message from CVS:
reviewed by: <delete if not using a buddy>
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_video_caps), (gst_riff_create_audio_caps),
(gst_riff_create_video_template_caps),
(gst_riff_create_audio_template_caps):
* gst-libs/gst/riff/riff-media.h:
* gst-libs/gst/riff/riff-read.c:
(gst_riff_read_strf_vids_with_data), (gst_riff_read_strf_vids):
* gst-libs/gst/riff/riff-read.h:
* gst/avi/gstavidemux.c: (gst_avi_demux_add_stream):
Add MS RLE support. I added some functions to read out strf chunks
into strf chunks and the data behind it. This is usually color
palettes (as in RLE, but also in 8-bit RGB). Also use those during
caps creation. Lastly, add ADPCM (similar to wavparse - which
should eventually be rifflib based).
* gst/matroska/matroska-demux.c: (gst_matroska_demux_class_init),
(gst_matroska_demux_init), (gst_matroska_demux_reset):
* gst/matroska/matroska-demux.h:
Remove placeholders for some prehistoric tagging system. Didn't add
support for any tag system really anyway.
* gst/qtdemux/qtdemux.c:
Add support for audio/x-m4a (MPEG-4) through spider.
* gst/wavparse/gstwavparse.c: (gst_wavparse_parse_fmt),
(gst_wavparse_loop):
ADPCM support (#135862). Increase max. buffer size because we
cannot split buffers for ADPCM (screws references) and I've seen
files with 2048 byte chunks. 4096 seems safe for now.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Fix typo in divxversion (3 instead of 4 for "DIVX" fourcc).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add MS video v1.
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
(gst_avi_demux_stream_data):
Add support for "rec-list" chunks.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Fix divx caps mismatch and move from video/x-jpeg to image/jpeg
so that MJPEG plays back.
Original commit message from CVS:
* gst-libs/gst/play/play.c (gst_play_audio_fixate)
(gst_play_video_fixate): Check so the structure has the field
before trying to fixate them, this makes it possible to have
fakesinks for video and audio output without printing errors on
the output console.
Original commit message from CVS:
2004-03-15 Julien MOUTTE <julien@moutte.net>
* gst-libs/gst/play/play.c: (gst_play_get_framerate),
(gst_play_get_sink_element): First draft of gst_play_get_framerate.
* gst-libs/gst/play/play.h:
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/gconf/Makefile.am:
* pkgconfig/Makefile.am:
move gstreamer-gconf pkgconfig files to pkgconfig/ dir. Make sure
they get rebuilt properly
* configure.ac:
when checking for vorbis, try pkgconfig first.
* gst/modplug/gstmodplug.cc:
add fixate function
Original commit message from CVS:
* gst-libs/gst/xoverlay/xoverlay.c: (gst_x_overlay_set_xwindow_id),
(gst_x_overlay_got_xwindow_id):
* gst-libs/gst/xoverlay/xoverlay.h:
replace XID with unsigned long to get rid of the xlibs dependency in
XOverlay (fixes#137004)
Original commit message from CVS:
* ext/lcs/Makefile.am: Fix so that the lcs colorspace plugin
doesn't conflict with the internal colorspace plugin.
* gst-libs/gst/audio/make_filter: Use `` instead of $() to
satisfy the crappy-ass shell shipped by a certain vendor.
* gst/videofilter/make_filter: same (bug #135299)
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_init), (gst_faad_srcgetcaps),
(gst_faad_chain): Fix negotiation.
* ext/librfb/gstrfbsrc.c: (gst_rfbsrc_handle_src_event): Add
key and button events.
* gst-libs/gst/floatcast/floatcast.h: Fix a minor bug in this
dung heap of code.
* gst-libs/gst/gconf/gstreamer-gconf-uninstalled.pc.in: gstgconf
depends on gconf
* gst-libs/gst/gconf/gstreamer-gconf.pc.in: same
* gst-libs/gst/play/play.c: (gst_play_pipeline_setup),
(gst_play_video_fixate), (gst_play_audio_fixate): Add a fixate
function to encourage better negotiation, particularly between
audioconvert and osssink.
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain):
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak): Make some debugging
more important.
* gst/typefind/gsttypefindfunctions.c: Fix mistake in flash
typefinding.
* gst/vbidec/vbiscreen.c: Add glib header
* pkgconfig/gstreamer-play.pc.in: Depends on gst-interfaces.
Original commit message from CVS:
2004-03-03 Julien MOUTTE <julien@moutte.net>
* gst-libs/gst/gconf/gconf.c: (gst_gconf_get_default_video_sink):
Using ximagesink as a default if no gconf key found. We should
probably consider using alsasink instead of osssink for the audio
part.
Original commit message from CVS:
2004-02-27 Benjamin Otte <otte@gnome.org>
* gst-libs/gst/audio/audio.h:
add macro to make sure header isn't included twice
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_process_chunk):
don't use gst_buffer_free
* gst/playondemand/filter.func:
don't usae gst_data_free. Free data only once.
Original commit message from CVS:
* gst-libs/gst/colorbalance/Makefile.am:
* gst-libs/gst/mixer/Makefile.am:
* gst-libs/gst/tuner/Makefile.am:
* gst/level/Makefile.am: -marshal.[ch] and -enum.[ch] files
should not be disted, -marshal.h files should not be installed,
and -enum.h files _should_ be installed. Fix to make this the
case.
Original commit message from CVS:
2004-02-15 Julien MOUTTE <julien@moutte.net>
* examples/gstplay/player.c: (got_eos), (main): Adding some
output for debugging.
* gst-libs/gst/play/play.c: (gst_play_state_change): Stop our
timeouts if we go to any state different from PLAYING.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek): Fix some
more EOS bugs in riff lib.
Original commit message from CVS:
2004-02-14 Julien MOUTTE <julien@moutte.net>
* gst-libs/gst/play/play.c: (gst_play_connect_visualization): Disable
visualization until i find a way to fix switch correctly.
* gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head): Fix a bug when
EOS arrives.
* gst/switch/gstswitch.c: (gst_switch_release_pad),
(gst_switch_request_new_pad), (gst_switch_poll_sinkpads),
(gst_switch_loop), (gst_switch_dispose), (gst_switch_class_init):
Reworked switch to get a more correct behaviour with events and refing
of data stored in sinkpads.
* gst/switch/gstswitch.h: Adding an eos flag for every sinkpad so that
we don't pull from a pad in EOS.
Original commit message from CVS:
2004-02-08 Benjamin Otte <in7y118@public.uni-hamburg.de>
* gst-libs/gst/colorbalance/Makefile.am:
* gst-libs/gst/navigation/Makefile.am:
* gst-libs/gst/xoverlay/Makefile.am:
remove unused GST_OPT_CFLAGS from Makefiles
include X_CFLAGS and X_LIBS in xoverlay. (#131948)
Original commit message from CVS:
2004-02-05 Benjamin Otte <in7y118@public.uni-hamburg.de>
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_info):
use correct GST_TAG_ENCODER tag
Original commit message from CVS:
2004-02-04 Julien MOUTTE <julien@moutte.net>
* gst-libs/gst/xoverlay/xoverlay.c: (gst_x_overlay_expose):
* gst-libs/gst/xoverlay/xoverlay.h: Adding the _expose method to tell
an overlay to redraw the image because it has been exposed.
* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_destroy),
(gst_ximagesink_ximage_put), (gst_ximagesink_expose),
(gst_ximagesink_xoverlay_init), (gst_ximagesink_init):
* sys/ximage/ximagesink.h: Implement expose method from XOverlay
interface
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_destroy),
(gst_xvimagesink_xvimage_put), (gst_xvimagesink_expose),
(gst_xvimagesink_xoverlay_init), (gst_xvimagesink_init):
* sys/xvimage/xvimagesink.h: Implement expose method from XOverlay
interface
Original commit message from CVS:
* gst-libs/gst/audio/.cvsignore:
Ignore generated file.
* gst-libs/gst/audio/Makefile.am:
Do not install example filter.
Original commit message from CVS:
2004-02-02 Julien MOUTTE <julien@moutte.net>
* gst-libs/gst/play/play.c: (gst_play_pipeline_setup),
(gst_play_set_audio_sink): Moving volume in the audio thread for
instantaneous volume change. Maybe i will add another volume in front
of visualization later, not sure yet though.
Original commit message from CVS:
2004-01-30 Ronald Bultje <rbultje@ronald.bitfreak.net>
* gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head),
(gst_riff_read_seek):
* gst/matroska/ebml-read.c: (gst_ebml_read_element_id),
(gst_ebml_read_seek):
Fix event handling.
Original commit message from CVS:
2004-01-29 Julien MOUTTE <julien@moutte.net>
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_get): Fixing seeking
emiting FLUSH and even before DISCONT.
* gst-libs/gst/play/gstplay.c: (gst_play_seek_to_time): Fix seeking to
get the best instant seeking as possible yay!
Original commit message from CVS:
2004-01-28 Julien MOUTTE <julien@moutte.net>
* gst-libs/gst/play/gstplay.c: (gst_play_seek_to_time): seek on video
sink element first.
* gst/videoscale/gstvideoscale.c:
(gst_videoscale_handle_src_event): Fixing src event handler.
Original commit message from CVS:
2004-01-27 Julien MOUTTE <julien@moutte.net>
* gst-libs/gst/play/gstplay.c: (gst_play_tick_callback),
(gst_play_seek_to_time): Fixing the way to get current position.
Original commit message from CVS:
2004-01-26 Benjamin Otte <in7y118@public.uni-hamburg.de>
* gst-libs/gst/audio/audio.h:
remove buffer-frames from audio caps
* gst/audioconvert/gstaudioconvert.c:
fix plugin to really work.
Original commit message from CVS:
2004-01-25 Ronald Bultje <rbultje@ronald.bitfreak.net>
* gst-libs/gst/mixer/mixer.c:
* gst-libs/gst/propertyprobe/propertyprobe.c:
* gst-libs/gst/tuner/tuner.c: (gst_tuner_find_norm_by_name),
(gst_tuner_find_channel_by_name):
* gst-libs/gst/tuner/tuner.h:
Add gtk-doc style comments. Also fix a function name.
Original commit message from CVS:
2004-01-25 Julien MOUTTE <julien@moutte.net>
* gst-libs/gst/play/gstplay.c: (gst_play_pipeline_setup),
(gst_play_identity_handoff), (gst_play_set_location),
(gst_play_set_visualization), (gst_play_connect_visualization): Another
try in visualization implementation. Still have an issue with switch
blocking when pulling from video_queue and only audio comes out of
spider.
* gst/switch/gstswitch.c: (gst_switch_release_pad),
(gst_switch_poll_sinkpads), (gst_switch_class_init): Implementing pad
release method. And check if the pad is usable before pulling.
Original commit message from CVS:
2004-01-25 Ronald Bultje <rbultje@ronald.bitfreak.net>
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_info):
Additional pad usability check.
* gst/mpeg1videoparse/gstmp1videoparse.c: (gst_mp1videoparse_init),
(mp1videoparse_find_next_gop), (gst_mp1videoparse_time_code),
(gst_mp1videoparse_real_chain):
Fix MPEG video stream parsing. The original plugin had several
issues, including not timestamping streams where the source was
not timestamped (this happens with PTS values in mpeg system
streams, but MPEG video is also a valid stream on its own so
that needs timestamps too). We use the display time code for that
for now. Also, if one incoming buffer contains multiple valid
frames, we push them all on correctly now, including proper EOS
handling. Lastly, several potential segfaults were fixed, and we
properly sync on new sequence/gop headers to include them in next,
not previous frames (since they're header for the next frame, not
the previous). Also see #119206.
* gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_chain),
(bpf_from_header):
Move caps setting so we only do it after finding several valid
MPEG-1 fraes sequentially, not right after the first one (which
might be coincidental).
* gst/typefind/gsttypefindfunctions.c: (mpeg1_sys_type_find),
(mpeg_video_type_find), (mpeg_video_stream_type_find),
(plugin_init):
Add unsynced MPEG video stream typefinding, and change some
probability values so we detect streams rightly. The idea is as
follows: I can have an unsynced system stream which contains
video. In the current code, I would randomly get a type for either
system or video stream type found, because the probabilities are
being calculated rather randomly. I now use fixed values, so we
always prefer system stream if that was found (and that is how it
should be). If no system stream was found, we can still identity
the stream as video-only.
