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)
