Robin Stocker [Sat, 8 Dec 2007 18:38:39 +0000 (18:38 +0000)]
gst/subparse/gstsubparse.c: Some .srt files start with chunk number 0 and not chunk number 1, recognise and accept th...
Original commit message from CVS:
Patch by: Robin Stocker <robin dot stocker at gmx dot ch>
* gst/subparse/gstsubparse.c: (gst_sub_parse_data_format_autodetect):
Some .srt files start with chunk number 0 and not chunk number 1,
recognise and accept those as well (fixes #502497).
* tests/check/elements/subparse.c: (srt_input), (srt_input0),
(test_src):
Add unit test for the above.
Wim Taymans [Thu, 6 Dec 2007 12:08:21 +0000 (12:08 +0000)]
gst/playback/gstplay-enum.*: Add missing files.
Original commit message from CVS:
* gst/playback/gstplay-enum.c:
(register_gst_autoplug_select_result),
(gst_autoplug_select_result_get_type):
* gst/playback/gstplay-enum.h:
Add missing files.
Wim Taymans [Wed, 5 Dec 2007 17:11:48 +0000 (17:11 +0000)]
gst/playback/Makefile.am: Group decodebin2 and uridecodebin into the same plugin so that they can share the GEnumType.
Original commit message from CVS:
* gst/playback/Makefile.am:
Group decodebin2 and uridecodebin into the same plugin so that they
can share the GEnumType.
* gst/playback/gstdecodebin2.c: (_gst_array_accumulator),
(_gst_select_accumulator), (gst_decode_bin_class_init),
(gst_decode_bin_init), (gst_decode_bin_autoplug_sort),
(gst_decode_bin_autoplug_select), (gst_decode_bin_autoplug_add),
(analyze_new_pad), (connect_pad), (gst_decode_bin_plugin_init):
Add signal to sort factories instead of the more awkward autoplug-select
signal.
Modify autoplug_select so that we can try, skip or expose the
autopluggin of an element on a pad.
* gst/playback/gstfactorylists.c: (compare_ranks),
(decoders_filter), (sinks_filter), (gst_factory_list_is_type),
(element_filter), (gst_factory_list_get_elements),
(gst_factory_list_debug), (gst_factory_list_filter):
* gst/playback/gstfactorylists.h:
Simplify the API, allow getting elements based on mask.
* gst/playback/gstplay-marshal.list:
Add some more marshallers.
* gst/playback/gstplaybin2.c: (init_group), (gst_play_bin_init),
(gst_play_bin_finalize), (pad_removed_cb), (autoplug_factories_cb),
(autoplug_select_cb), (activate_group):
Add support for managing non-raw sinks by providing a custom element and
sink list to decodebin2.
Try to plug non-raw sinks when decodebin2 using autoplug-select of
decodebin2.
* gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
(gst_play_sink_set_mode), (gst_play_sink_request_pad):
* gst/playback/gstplaysink.h:
Add support for raw and non-raw sinks.
Add support to force sinks selected by playbin2.
Don't plug raw converters for non-raw sinks.
* gst/playback/gsturidecodebin.c: (_gst_array_accumulator),
(_gst_select_accumulator), (gst_uri_decode_bin_class_init),
(proxy_autoplug_select_signal), (gst_uri_decode_bin_plugin_init),
(plugin_init):
Use right accumulators.
Proxy new signal.
Wim Taymans [Mon, 3 Dec 2007 13:47:00 +0000 (13:47 +0000)]
gst-libs/gst/rtp/gstbasertppayload.c: Use runnning time as the base time instead of the timestamp.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
Use runnning time as the base time instead of the timestamp.
Spotted by Saur on IRC.
Edward Hervey [Mon, 3 Dec 2007 11:32:30 +0000 (11:32 +0000)]
gst-libs/gst/riff/riff-media.c: Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
Wim Taymans [Mon, 3 Dec 2007 10:58:14 +0000 (10:58 +0000)]
ext/ogg/gstoggdemux.c: If we find a new serial number but it does not contain a BOS page, make sure we initialize the...
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_bisect_forward_serialno),
(gst_ogg_demux_read_chain):
If we find a new serial number but it does not contain a BOS page, make
sure we initialize the chain to NULL because else we will try to scan it
and crash. Fixes #500763
Wim Taymans [Fri, 30 Nov 2007 17:47:15 +0000 (17:47 +0000)]
gst/playback/: Refactor some common code to filter factories and check caps compat.
Original commit message from CVS:
* gst/playback/Makefile.am:
* gst/playback/gstfactorylists.c: (compare_ranks), (print_feature),
(get_feature_array), (decoders_filter), (sinks_filter),
(gst_factory_list_get_decoders), (gst_factory_list_get_sinks),
(gst_factory_list_filter):
* gst/playback/gstfactorylists.h:
Refactor some common code to filter factories and check caps compat.
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
(gst_decode_bin_init), (gst_decode_bin_dispose),
(gst_decode_bin_autoplug_continue),
(gst_decode_bin_autoplug_factories),
(gst_decode_bin_autoplug_select), (analyze_new_pad),
(find_compatibles):
* gst/playback/gstplaybin.c:
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(gst_play_bin_init), (gst_play_bin_finalize),
(autoplug_factories_cb), (activate_group):
* gst/playback/gstqueue2.c:
* gst/playback/gsturidecodebin.c: (proxy_unknown_type_signal),
(proxy_autoplug_continue_signal),
(proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
(proxy_drained_signal):
Add some more debug info and use factor filtering code.
Julien Moutte [Mon, 26 Nov 2007 13:19:46 +0000 (13:19 +0000)]
configure.ac: Add QuickTime Wrapper plug-in.
Original commit message from CVS:
2007-11-26 Julien Moutte <julien@fluendo.com>
* configure.ac: Add QuickTime Wrapper plug-in.
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_push_drain), (gst_speex_resample_process): Fix
build on Mac OS X Leopard. Incorrect printf format arguments.
* sys/Makefile.am:
* sys/qtwrapper/Makefile.am:
* sys/qtwrapper/audiodecoders.c:
(qtwrapper_audio_decoder_base_init),
(qtwrapper_audio_decoder_class_init),
(qtwrapper_audio_decoder_init),
(clear_AudioStreamBasicDescription), (fill_indesc_mp3),
(fill_indesc_aac), (fill_indesc_samr), (fill_indesc_generic),
(make_samr_magic_cookie), (open_decoder),
(qtwrapper_audio_decoder_sink_setcaps), (process_buffer_cb),
(qtwrapper_audio_decoder_chain),
(qtwrapper_audio_decoder_sink_event),
(qtwrapper_audio_decoders_register):
* sys/qtwrapper/codecmapping.c: (audio_caps_from_string),
(fourcc_to_caps):
* sys/qtwrapper/codecmapping.h:
* sys/qtwrapper/imagedescription.c: (image_description_for_avc1),
(image_description_for_mp4v), (image_description_from_stsd_buffer),
(image_description_from_codec_data):
* sys/qtwrapper/imagedescription.h:
* sys/qtwrapper/qtutils.c: (get_name_info_from_component),
(get_output_info_from_component), (dump_avcc_atom),
(dump_image_description), (dump_codec_decompress_params),
(addSInt32ToDictionary), (dump_cvpixel_buffer),
(DestroyAudioBufferList), (AllocateAudioBufferList):
* sys/qtwrapper/qtutils.h:
* sys/qtwrapper/qtwrapper.c: (plugin_init):
* sys/qtwrapper/qtwrapper.h:
* sys/qtwrapper/videodecoders.c:
(qtwrapper_video_decoder_base_init),
(qtwrapper_video_decoder_class_init),
(qtwrapper_video_decoder_init), (qtwrapper_video_decoder_finalize),
(fill_image_description), (new_image_description), (close_decoder),
(open_decoder), (qtwrapper_video_decoder_sink_setcaps),
(decompressCb), (qtwrapper_video_decoder_chain),
(qtwrapper_video_decoder_sink_event),
(qtwrapper_video_decoders_register): Initial import of QuickTime
wrapper jointly developped by Songbird authors (Pioneers of the
Inevitable) and Fluendo.
Stefan Kost [Mon, 26 Nov 2007 12:25:55 +0000 (12:25 +0000)]
gst/: Add GAP-flag support.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/volume/gstvolume.c:
* gst/volume/gstvolume.h:
Add GAP-flag support.
Sebastian Dröge [Mon, 26 Nov 2007 08:43:25 +0000 (08:43 +0000)]
gst/speexresample/: Update speex resampler to latest SVN. We're now down to only the changes noted in README again.
Original commit message from CVS:
* gst/speexresample/README:
* gst/speexresample/arch.h:
* gst/speexresample/resample.c: (resampler_basic_direct_single),
(resampler_basic_direct_double),
(resampler_basic_interpolate_single),
(resampler_basic_interpolate_double),
(speex_resampler_process_native), (speex_resampler_process_float),
(speex_resampler_process_int),
(speex_resampler_process_interleaved_float),
(speex_resampler_process_interleaved_int),
(speex_resampler_get_input_latency),
(speex_resampler_get_output_latency):
* gst/speexresample/speex_resampler.h:
Update speex resampler to latest SVN. We're now down to only the
changes noted in README again.
* gst/speexresample/speex_resampler_wrapper.h:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_push_drain), (gst_speex_resample_query):
Adjust to API changes.
Julien Moutte [Sat, 24 Nov 2007 15:02:01 +0000 (15:02 +0000)]
tests/examples/seek/seek.c: Increase the range of the rate selector as I would like to test QOS behavior at higher fo...
Original commit message from CVS:
2007-11-24 Julien MOUTTE <julien@moutte.net>
* tests/examples/seek/seek.c: (main): Increase the range of the
rate selector as I would like to test QOS behavior at higher
forward and reverse playback speed like say 64x.
Sebastian Dröge [Fri, 23 Nov 2007 10:21:31 +0000 (10:21 +0000)]
gst/speexresample/gstspeexresample.c: Only post the latency message if we have a resampler state already.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_update_state):
Only post the latency message if we have a resampler state already.
Sebastian Dröge [Fri, 23 Nov 2007 10:21:11 +0000 (10:21 +0000)]
gst/audioresample/gstaudioresample.c: Implement latency query.
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
(audioresample_query), (audioresample_query_type),
(gst_audioresample_set_property):
Implement latency query.
Sebastian Dröge [Fri, 23 Nov 2007 10:01:33 +0000 (10:01 +0000)]
gst/speexresample/gstspeexresample.c: Also post GST_MESSAGE_LATENCY if the latency changes.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_update_state):
Also post GST_MESSAGE_LATENCY if the latency changes.
Sebastian Dröge [Fri, 23 Nov 2007 08:48:50 +0000 (08:48 +0000)]
gst/speexresample/: Add functions to push the remaining samples and to get the latency of the resampler. These will g...
Original commit message from CVS:
* gst/speexresample/resample.c: (speex_resampler_get_latency),
(speex_resampler_drain_float), (speex_resampler_drain_int),
(speex_resampler_drain_interleaved_float),
(speex_resampler_drain_interleaved_int):
* gst/speexresample/speex_resampler.h:
* gst/speexresample/speex_resampler_wrapper.h:
Add functions to push the remaining samples and to get the latency
of the resampler. These will get added to Speex SVN in this or a
slightly changed form at some point too and should get merged then
again.
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_init),
(gst_speex_resample_init_state),
(gst_speex_resample_transform_size),
(gst_speex_resample_push_drain), (gst_speex_resample_event),
(gst_speex_fix_output_buffer), (gst_speex_resample_process),
(gst_speex_resample_query), (gst_speex_resample_query_type):
Drop the prepending zeroes and output the remaining samples on EOS.
Also properly implement the latency query for this. speexresample
should be completely ready for production use now.
Wim Taymans [Wed, 21 Nov 2007 18:02:21 +0000 (18:02 +0000)]
gst-libs/gst/audio/gstbaseaudiosink.c: Our EOS time contains the base_time, _wait_eos() expects a running_time so we ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain):
Our EOS time contains the base_time, _wait_eos() expects a running_time
so we have to subtract the base_time again before calling the function.
This fixes an EOS regression where the base_time was added twice and EOS
took longer and longer in certain situations.
Fixes #498767.
Wim Taymans [Wed, 21 Nov 2007 13:04:17 +0000 (13:04 +0000)]
Expose methods for some object properties so that subclasses can more easily configure them.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
(gst_base_audio_sink_set_provide_clock),
(gst_base_audio_sink_get_provide_clock),
(gst_base_audio_sink_set_slave_method),
(gst_base_audio_sink_get_slave_method),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_drain),
(gst_base_audio_sink_none_slaving),
(gst_base_audio_sink_handle_slaving):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Expose methods for some object properties so that subclasses can more
easily configure them.
Added slave method none, that completely disables slaving to the
internal clock.
API: gst_base_audio_sink_set_provide_clock()
API: gst_base_audio_sink_get_provide_clock()
API: gst_base_audio_sink_set_slave_method()
API: gst_base_audio_sink_get_slave_method()
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_set_provide_clock),
(gst_base_audio_src_get_provide_clock),
(gst_base_audio_src_set_property),
(gst_base_audio_src_get_property), (gst_base_audio_src_create):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
Expose methods for some object properties so that subclasses can more
easily configure them.
API: gst_base_audio_src_set_provide_clock()
API: gst_base_audio_src_get_provide_clock()
Sebastian Dröge [Wed, 21 Nov 2007 10:18:56 +0000 (10:18 +0000)]
gst/speexresample/README: Add README explaining where the resampling code was taken from and which changes were done.
Original commit message from CVS:
* gst/speexresample/README:
Add README explaining where the resampling code was taken from
and which changes were done.
* gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
(speex_free):
Use g_malloc() and friends instead of malloc() to achieve higher
portability and define the functions inline.
* gst/speexresample/speex_resampler.h:
Add back some useless preprocessor stuff to keep the diff between
our version and the one from the Speex SVN repository lower.
Sebastian Dröge [Tue, 20 Nov 2007 20:23:25 +0000 (20:23 +0000)]
gst/speexresample/gstspeexresample.c: Some small cleanup and addition of a TODO item.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_fix_output_buffer), (gst_speex_resample_transform):
Some small cleanup and addition of a TODO item.
Sebastian Dröge [Tue, 20 Nov 2007 12:56:00 +0000 (12:56 +0000)]
gst/speexresample/Makefile.am: Add missing file.
Original commit message from CVS:
* gst/speexresample/Makefile.am:
Add missing file.
Joe Peterson [Tue, 20 Nov 2007 07:53:56 +0000 (07:53 +0000)]
gst-libs/gst/sdp/gstsdpmessage.c: Fix compilation on FreeBSD (Gentoo). Fixes #498228.
Original commit message from CVS:
Patch by: Joe Peterson <lavajoe at gentoo dot org>
* gst-libs/gst/sdp/gstsdpmessage.c:
Fix compilation on FreeBSD (Gentoo). Fixes #498228.
Sebastian Dröge [Tue, 20 Nov 2007 07:47:27 +0000 (07:47 +0000)]
Add speexresample to the docs and while at that do a make update.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-bz2.xml:
* docs/plugins/inspect/plugin-cdxaparse.xml:
* docs/plugins/inspect/plugin-dtsdec.xml:
* docs/plugins/inspect/plugin-equalizer.xml:
* docs/plugins/inspect/plugin-faac.xml:
* docs/plugins/inspect/plugin-faad.xml:
* docs/plugins/inspect/plugin-filter.xml:
* docs/plugins/inspect/plugin-freeze.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-gsm.xml:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
* docs/plugins/inspect/plugin-h264parse.xml:
* docs/plugins/inspect/plugin-modplug.xml:
* docs/plugins/inspect/plugin-mpeg2enc.xml:
* docs/plugins/inspect/plugin-musepack.xml:
* docs/plugins/inspect/plugin-musicbrainz.xml:
* docs/plugins/inspect/plugin-nsfdec.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* docs/plugins/inspect/plugin-soundtouch.xml:
* docs/plugins/inspect/plugin-spcdec.xml:
* docs/plugins/inspect/plugin-spectrum.xml:
* docs/plugins/inspect/plugin-speed.xml:
* docs/plugins/inspect/plugin-tta.xml:
* docs/plugins/inspect/plugin-videosignal.xml:
* docs/plugins/inspect/plugin-xingheader.xml:
* docs/plugins/inspect/plugin-xvid.xml:
* gst/speexresample/gstspeexresample.h:
Add speexresample to the docs and while at that do a make update.
Sebastian Dröge [Tue, 20 Nov 2007 07:30:30 +0000 (07:30 +0000)]
gst/speexresample/gstspeexresample.c: If the resampler gives less output samples than expected adjust the output buff...
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_fix_output_buffer), (gst_speex_resample_process):
If the resampler gives less output samples than expected
adjust the output buffer and print a warning.
Sebastian Dröge [Tue, 20 Nov 2007 07:02:45 +0000 (07:02 +0000)]
Add resample element based on the Speex resampling algorithm.
Original commit message from CVS:
* configure.ac:
* gst/speexresample/arch.h:
* gst/speexresample/fixed_generic.h:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_base_init), (gst_speex_resample_class_init),
(gst_speex_resample_init), (gst_speex_resample_start),
(gst_speex_resample_stop), (gst_speex_resample_get_unit_size),
(gst_speex_resample_transform_caps),
(gst_speex_resample_init_state), (gst_speex_resample_update_state),
(gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
(gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
(gst_speex_resample_event), (gst_speex_resample_check_discont),
(gst_speex_resample_process), (gst_speex_resample_transform),
(gst_speex_resample_set_property),
(gst_speex_resample_get_property), (plugin_init):
* gst/speexresample/gstspeexresample.h:
* gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
(speex_free), (compute_func), (main), (sinc), (cubic_coef),
(resampler_basic_direct_single), (resampler_basic_direct_double),
(resampler_basic_interpolate_single),
(resampler_basic_interpolate_double), (update_filter),
(speex_resampler_init), (speex_resampler_init_frac),
(speex_resampler_destroy), (speex_resampler_process_native),
(speex_resampler_process_float), (speex_resampler_process_int),
(speex_resampler_process_interleaved_float),
(speex_resampler_process_interleaved_int),
(speex_resampler_set_rate), (speex_resampler_get_rate),
(speex_resampler_set_rate_frac), (speex_resampler_get_ratio),
(speex_resampler_set_quality), (speex_resampler_get_quality),
(speex_resampler_set_input_stride),
(speex_resampler_get_input_stride),
(speex_resampler_set_output_stride),
(speex_resampler_get_output_stride), (speex_resampler_skip_zeros),
(speex_resampler_reset_mem), (speex_resampler_strerror):
* gst/speexresample/speex_resampler.h:
* gst/speexresample/speex_resampler_float.c:
* gst/speexresample/speex_resampler_int.c:
* gst/speexresample/speex_resampler_wrapper.h:
Add resample element based on the Speex resampling algorithm.
Sebastian Dröge [Mon, 19 Nov 2007 12:30:22 +0000 (12:30 +0000)]
tests/check/libs/fft.c: Fix scaling to really have dB instead of something else.
Original commit message from CVS:
* tests/check/libs/fft.c: (GST_START_TEST):
Fix scaling to really have dB instead of something else.
Julien Moutte [Mon, 19 Nov 2007 12:08:16 +0000 (12:08 +0000)]
tests/examples/seek/seek.c: There's a nice macro to check
Original commit message from CVS:
2007-11-19 Julien MOUTTE <julien@moutte.net>
* tests/examples/seek/seek.c: (main): There's a nice macro to
check
GTK version, use it.
Julien Moutte [Mon, 19 Nov 2007 11:59:20 +0000 (11:59 +0000)]
tests/examples/seek/seek.c: Try to support stable version of GTK.
Original commit message from CVS:
2007-11-19 Julien MOUTTE <julien@moutte.net>
* tests/examples/seek/seek.c: (main): Try to support stable version
of GTK.
Stefan Kost [Sat, 17 Nov 2007 15:25:15 +0000 (15:25 +0000)]
gst/playback/: Fix the build + little README update.
Original commit message from CVS:
* gst/playback/README:
* gst/playback/test7.c:
Fix the build + little README update.
Wim Taymans [Fri, 16 Nov 2007 16:02:45 +0000 (16:02 +0000)]
tests/examples/seek/seek.c: Add playbin2 seek pipeline.
Original commit message from CVS:
* tests/examples/seek/seek.c: (make_playerbin2_pipeline), (main):
Add playbin2 seek pipeline.
Wim Taymans [Fri, 16 Nov 2007 15:44:48 +0000 (15:44 +0000)]
gst/playback/: Add playbin2.
Original commit message from CVS:
* gst/playback/Makefile.am:
* gst/playback/gstplayback.c: (plugin_init):
* gst/playback/test7.c: (update_scale), (warning_cb), (error_cb),
(eos_cb), (about_to_finish_cb), (main):
Add playbin2.
Added gapless playback example.
* gst/playback/gstplaybasebin.c:
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (gst_play_bin_plugin_init):
* gst/playback/gstqueue2.c:
* gst/playback/test.c:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
(pad_removed_cb):
* gst/playback/gststreaminfo.h:
Change email.
* gst/playback/gstplaybin2.c: (gst_play_bin_get_type),
(gst_play_bin_class_init), (init_group), (gst_play_bin_init),
(gst_play_bin_dispose), (gst_play_bin_set_uri),
(gst_play_bin_set_suburi), (gst_play_bin_set_property),
(gst_play_bin_get_property), (gst_play_bin_handle_message),
(pad_added_cb), (pad_removed_cb), (no_more_pads_cb), (perform_eos),
(drained_cb), (unlink_group), (activate_group),
(setup_next_source), (gst_play_bin_change_state),
(gst_play_bin2_plugin_init):
Added raw first version of playbin2. Does chained oggs and gapless
playback fine. No support for raw sinks yet. No visualisations or
subtitles yet.
* gst/playback/gstplaysink.c: (gst_play_sink_get_type),
(gst_play_sink_class_init), (gst_play_sink_init),
(gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
(gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
(gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
(gst_play_sink_set_property), (gst_play_sink_get_property),
(post_missing_element_message), (free_chain), (add_chain),
(activate_chain), (gen_video_chain), (gen_text_element),
(gen_audio_chain), (gen_vis_element), (gst_play_sink_get_mode),
(gst_play_sink_set_mode), (gst_play_sink_request_pad),
(gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
(gst_play_sink_send_event), (gst_play_sink_change_state):
* gst/playback/gstplaysink.h:
Added Element that abstracts the sinks and their pipelines for playbin2.
Wim Taymans [Fri, 16 Nov 2007 15:05:07 +0000 (15:05 +0000)]
gst/playback/gststreamselector.*: Improve streamselector, make it select and unselect the current pad more intelligen...
Original commit message from CVS:
* gst/playback/gststreamselector.c: (gst_selector_pad_get_type),
(gst_selector_pad_class_init), (gst_selector_pad_init),
(gst_selector_pad_finalize), (gst_selector_pad_reset),
(gst_selector_pad_get_linked_pads), (gst_selector_pad_event),
(gst_selector_pad_getcaps), (gst_selector_pad_bufferalloc),
(gst_selector_pad_chain), (gst_stream_selector_get_type),
(gst_stream_selector_base_init), (gst_stream_selector_class_init),
(gst_stream_selector_init), (gst_stream_selector_set_property),
(gst_stream_selector_get_linked_pad),
(gst_stream_selector_getcaps),
(gst_stream_selector_is_active_sinkpad),
(gst_stream_selector_activate_sinkpad),
(gst_stream_selector_get_linked_pads),
(gst_stream_selector_request_new_pad),
(gst_stream_selector_release_pad):
* gst/playback/gststreamselector.h:
Improve streamselector, make it select and unselect the current pad more
intelligently.
Subclass GstPad for the sinkpads of the selector.
Handle segments more correctly.
Fix caps negotiation.
Implement release_pad.
Wim Taymans [Fri, 16 Nov 2007 12:51:44 +0000 (12:51 +0000)]
gst/playback/gstdecodebin2.c: Add drained signal fired when decodebin finishes decoding the data.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
(gst_decode_group_check_if_drained), (source_pad_event_probe),
(remove_fakesink):
Add drained signal fired when decodebin finishes decoding the data.
Remove deprecated STATE_DIRTY message.
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
(unknown_type_cb), (new_decoded_pad_cb), (pad_removed_cb),
(analyse_source), (proxy_drained_signal), (make_decoder),
(source_new_pad), (value_list_append_structure_list),
(handle_redirect_message), (handle_message):
Proxy the new drained signal.
Handle pad removed from decodebin.
Handle redirect messages by sorting multiple redirections based on the
connection speed.
Tommi Myöhänen [Fri, 16 Nov 2007 11:22:09 +0000 (11:22 +0000)]
gst-libs/gst/rtsp/gstrtspmessage.c: Fix leaking headers. Fixes #496761.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
Fix leaking headers. Fixes #496761.
Tommi Myöhänen [Fri, 16 Nov 2007 11:16:58 +0000 (11:16 +0000)]
sys/: Don't leak the PAR on errors. Fixes #496731.
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get),
(gst_ximagesink_change_state):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get):
Don't leak the PAR on errors. Fixes #496731.
Tim-Philipp Müller [Fri, 16 Nov 2007 10:14:34 +0000 (10:14 +0000)]
gst-libs/gst/tag/gstid3tag.c: Add mapping for audio cd discid tags, so we can extract them from tags as well (see #34...
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c: (user_tag_matches),
(gst_tag_from_id3_user_tag):
Add mapping for audio cd discid tags, so we can extract
them from tags as well (see #347848). Also compare identifiers
in ID3v2 TXXX frames in a case-insensitive way to increase
compatibility when reading tags (discid vs. DiscID vs. DiscId).
Jan Schmidt [Fri, 16 Nov 2007 01:21:40 +0000 (01:21 +0000)]
gst-plugins-base.doap: Oops, fix the release name.
Original commit message from CVS:
* gst-plugins-base.doap:
Oops, fix the release name.
Jan Schmidt [Fri, 16 Nov 2007 00:44:58 +0000 (00:44 +0000)]
gst-plugins-base.doap: Add 0.10.15 release
Original commit message from CVS:
* gst-plugins-base.doap:
Add 0.10.15 release
Jan Schmidt [Fri, 16 Nov 2007 00:24:55 +0000 (00:24 +0000)]
configure.ac: Back to CVS
Original commit message from CVS:
* configure.ac:
Back to CVS
Jan Schmidt [Fri, 16 Nov 2007 00:14:33 +0000 (00:14 +0000)]
configure.ac: releasing 0.10.15, "No need to argue"
Original commit message from CVS:
=== release 0.10.15 ===
2007-11-15 Jan Schmidt <jan.schmidt@sun.com>
* configure.ac:
releasing 0.10.15, "No need to argue"
Jan Schmidt [Fri, 16 Nov 2007 00:04:24 +0000 (00:04 +0000)]
Update .po files
Original commit message from CVS:
Update .po files
Jan Schmidt [Thu, 15 Nov 2007 21:40:53 +0000 (21:40 +0000)]
win32/vs6/libgstfft.dsp: Convert line endings to DOS.
Original commit message from CVS:
* win32/vs6/libgstfft.dsp:
Convert line endings to DOS.
Sébastien Moutte [Thu, 15 Nov 2007 21:14:04 +0000 (21:14 +0000)]
win32/: Add a project file for fft plugin and remove socket based plugin which don't build from the workspace.* win32...
Original commit message from CVS:
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstfft.dsp:
* win32/MANIFEST:
Add a project file for fft plugin and remove socket
based plugin which don't build from the workspace.* win32/vs6/libgstaudio.dsp:
* win32/vs6/libgstrtp.dsp:
* win32/vs6/libgsttag.dsp:
Convert line endings back to DOS.
Fixes #496724
Jan Schmidt [Wed, 14 Nov 2007 12:27:13 +0000 (12:27 +0000)]
win32/vs6/: Convert line endings back to DOS
Original commit message from CVS:
* win32/vs6/libgstinterfaces.dsp:
* win32/vs6/libgstrtsp.dsp:
Convert line endings back to DOS
Jan Schmidt [Wed, 14 Nov 2007 11:08:48 +0000 (11:08 +0000)]
gst-libs/gst/fft/: Don't include malloc.h which doesn't exist on Mac OSX.
Original commit message from CVS:
* gst-libs/gst/fft/kiss_fft_f32.h:
* gst-libs/gst/fft/kiss_fft_f64.h:
* gst-libs/gst/fft/kiss_fft_s16.h:
* gst-libs/gst/fft/kiss_fft_s32.h:
Don't include malloc.h which doesn't exist on Mac OSX.
Instead, pull in glib.h and use g_malloc/g_free for
consistency. Fixes: #496548
Stefan Kost [Fri, 9 Nov 2007 15:54:45 +0000 (15:54 +0000)]
gst/playback/gstdecodebin2.c: Dont leak ghostpad. Fixes #475451.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
Dont leak ghostpad. Fixes #475451.
Wim Taymans [Fri, 9 Nov 2007 12:21:52 +0000 (12:21 +0000)]
Update some more docs and comments.
Original commit message from CVS:
* docs/design/design-decodebin.txt:
* gst/playback/gstdecodebin2.c: (analyze_new_pad):
Update some more docs and comments.
Sebastian Dröge [Wed, 7 Nov 2007 16:47:32 +0000 (16:47 +0000)]
Require GIO >= 0.1.2 and adjust unit test for an API change.
Original commit message from CVS:
* configure.ac:
* tests/check/pipelines/gio.c: (GST_START_TEST):
Require GIO >= 0.1.2 and adjust unit test for an API change.
Sebastian Dröge [Wed, 7 Nov 2007 15:18:54 +0000 (15:18 +0000)]
ext/gio/gstgio.h: Add macro to check if a stream supports seeking.
Original commit message from CVS:
* ext/gio/gstgio.h:
Add macro to check if a stream supports seeking.
* ext/gio/Makefile.am:
* ext/gio/gstgiobasesink.c: (gst_gio_base_sink_base_init),
(gst_gio_base_sink_class_init), (gst_gio_base_sink_init),
(gst_gio_base_sink_finalize), (gst_gio_base_sink_start),
(gst_gio_base_sink_stop), (gst_gio_base_sink_unlock),
(gst_gio_base_sink_unlock_stop), (gst_gio_base_sink_event),
(gst_gio_base_sink_render), (gst_gio_base_sink_query),
(gst_gio_base_sink_set_stream):
* ext/gio/gstgiobasesink.h:
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_base_init),
(gst_gio_base_src_class_init), (gst_gio_base_src_init),
(gst_gio_base_src_finalize), (gst_gio_base_src_start),
(gst_gio_base_src_stop), (gst_gio_base_src_get_size),
(gst_gio_base_src_is_seekable), (gst_gio_base_src_unlock),
(gst_gio_base_src_unlock_stop), (gst_gio_base_src_check_get_range),
(gst_gio_base_src_create), (gst_gio_base_src_set_stream):
* ext/gio/gstgiobasesrc.h:
Refactor common GIO functions to GstGioBaseSink and GstGioBaseSrc
base classes that only require a GInputStream or GOutputStream to
work.
* ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
(gst_gio_sink_class_init), (gst_gio_sink_init),
(gst_gio_sink_finalize), (gst_gio_sink_start):
* ext/gio/gstgiosink.h:
* ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
(gst_gio_src_class_init), (gst_gio_src_init),
(gst_gio_src_finalize), (gst_gio_src_start):
* ext/gio/gstgiosrc.h:
Use the newly created base classes here.
* ext/gio/gstgio.c: (plugin_init):
* ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_base_init),
(gst_gio_stream_sink_class_init), (gst_gio_stream_sink_init),
(gst_gio_stream_sink_finalize), (gst_gio_stream_sink_set_property),
(gst_gio_stream_sink_get_property):
* ext/gio/gstgiostreamsink.h:
* ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_base_init),
(gst_gio_stream_src_class_init), (gst_gio_stream_src_init),
(gst_gio_stream_src_finalize), (gst_gio_stream_src_set_property),
(gst_gio_stream_src_get_property):
* ext/gio/gstgiostreamsrc.h:
Implement GstGioStreamSink and GstGioStreamSrc that have a property
to set the GInputStream/GOutputStream that should be used.
* tests/check/Makefile.am:
* tests/check/pipelines/.cvsignore:
* tests/check/pipelines/gio.c: (message_handler), (GST_START_TEST),
(gio_testsuite), (main):
Add unit test for giostreamsrc and giostreamsink.
Sebastian Dröge [Wed, 7 Nov 2007 11:48:09 +0000 (11:48 +0000)]
ext/gio/gstgio.c: Remove nowadays unnecessary workaround for a crash.
Original commit message from CVS:
* ext/gio/gstgio.c: (plugin_init):
Remove nowadays unnecessary workaround for a crash.
* ext/gio/gstgiosink.c: (gst_gio_sink_finalize),
(gst_gio_sink_start), (gst_gio_sink_stop),
(gst_gio_sink_unlock_stop):
* ext/gio/gstgiosink.h:
* ext/gio/gstgiosrc.c: (gst_gio_src_finalize), (gst_gio_src_start),
(gst_gio_src_stop), (gst_gio_src_unlock_stop):
* ext/gio/gstgiosrc.h:
Make the finalize function safer, clean up everything that could stay
around.
Reset the cancellable instead of creating a new one after cancelling
some operation.
Don't store the GFile in the element, it's only necessary for creating
the streams.
Sebastien Moutte [Tue, 6 Nov 2007 23:35:39 +0000 (23:35 +0000)]
gst-libs/gst/rtp/: Fix some C99-isms and and a missing function that some versions of
Original commit message from CVS:
Patch by: Sebastien Moutte <sebastien moutte net>
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
(gst_rtcp_unix_to_ntp):
* gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_name):
Fix some C99-isms and and a missing function that some versions of
MSVC don't like too much (#494346).
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstaudio.dsp:
* win32/vs6/libgstrtp.dsp:
* win32/vs6/libgsttag.dsp:
Update vs6 projects files (#494346).
Ole André Vadla Ravnås [Tue, 6 Nov 2007 16:38:49 +0000 (16:38 +0000)]
win32/common/: More missing symbols to export (fixes #493986).
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* win32/common/libgstaudio.def:
* win32/common/libgstcdda.def:
* win32/common/libgstinterfaces.def:
* win32/common/libgstnetbuffer.def:
* win32/common/libgstpbutils.def:
* win32/common/libgstrtp.def:
* win32/common/libgstrtsp.def:
* win32/common/libgsttag.def:
* win32/common/libgstvideo.def:
More missing symbols to export (fixes #493986).
Sebastian Dröge [Tue, 6 Nov 2007 11:58:59 +0000 (11:58 +0000)]
Remove the magnitude and phase calculation functions as these have very special use cases and can't even be used for ...
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/fft/gstfftf32.c:
* gst-libs/gst/fft/gstfftf32.h:
* gst-libs/gst/fft/gstfftf64.c:
* gst-libs/gst/fft/gstfftf64.h:
* gst-libs/gst/fft/gstffts16.c:
* gst-libs/gst/fft/gstffts16.h:
* gst-libs/gst/fft/gstffts32.c:
* gst-libs/gst/fft/gstffts32.h:
* tests/check/libs/fft.c: (GST_START_TEST):
Remove the magnitude and phase calculation functions as these have
very special use cases and can't even be used for the spectrum
element. Also adjust the docs to mention some properties of the used
FFT implemention, i.e. how the values are scaled. Fixes #492098.
Tim-Philipp Müller [Tue, 6 Nov 2007 11:09:30 +0000 (11:09 +0000)]
gst/playback/gstplaybasebin.c: Avoid crash when there are external subtitles (fixes #491722).
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (queue_threshold_reached),
(finish_source):
Avoid crash when there are external subtitles (fixes #491722).
Tim-Philipp Müller [Sat, 3 Nov 2007 10:39:21 +0000 (10:39 +0000)]
ext/alsa/: 'Could not open resource for writing' is not an acceptable even less so when we're trying to open it to re...
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_open):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_open):
'Could not open resource for writing' is not an acceptable
error message when we can't open the audio device (see #492334),
even less so when we're trying to open it to record something.
Ole André Vadla Ravnås [Fri, 2 Nov 2007 21:03:01 +0000 (21:03 +0000)]
win32/common/libgstrtp.def: Add some more missing symbols (#492813).
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* win32/common/libgstrtp.def:
Add some more missing symbols (#492813).
Thijs Vermeir [Fri, 2 Nov 2007 14:59:06 +0000 (14:59 +0000)]
tests/check/elements/audioconvert.c: Add check to make sure that the out caps have a channel layout set on them where...
Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
* tests/check/elements/audioconvert.c: (verify_convert):
Add check to make sure that the out caps have a channel layout
set on them where they should have one.
Vincent Torri [Thu, 1 Nov 2007 13:28:59 +0000 (13:28 +0000)]
gst-libs/gst/fft/: Include our own _stdint.h instead of sys/types.h, makes MingW happy (#492306).
Original commit message from CVS:
Patch by: Vincent Torri <vtorri at univ-evry dot fr>
* gst-libs/gst/fft/kiss_fft_s16.h: (KISS_FFT_S16_MALLOC):
* gst-libs/gst/fft/kiss_fft_s32.h: (KISS_FFT_S32_MALLOC):
Include our own _stdint.h instead of sys/types.h, makes MingW happy
(#492306).
* gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_create):
Use _pipe directly, GLib doesn't have a pipe() macro any longer
(it disappeared in GLib 2.14.0) (#492306).
* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/sdp/gstsdpmessage.c:
Fix includes and LIBS for win32/Mingw (#492306).
* tests/examples/dynamic/addstream.c (pause_play_stream):
Use more portable g_usleep() instead of sleep() (#492306).
Ole André Vadla Ravnås [Thu, 1 Nov 2007 12:51:57 +0000 (12:51 +0000)]
gst-libs/gst/audio/gstringbuffer.c: Return NULL instead of an enum that happens to be 0, fixes warning on MSVC (#4921...
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
(gst_ring_buffer_parse_caps):
Return NULL instead of an enum that happens to be 0, fixes warning
on MSVC (#492114).
* gst-libs/gst/audio/gstringbuffer.h:
No trailing commas in enum list (for gcc-2.9x).
* gst/videotestsrc/videotestsrc.c: (random_char):
Make information loss explicit instead of implicitly truncating to
eight bits via the return value. Fixes runtime error on MSVC when
using the debug CRT (#492114).
* win32/common/config.h.in:
Fix a bunch of '#undef FOO bar', which MSVC doesn't like (#492114).
* win32/common/libgstinterfaces.def:
* win32/common/libgstrtp.def:
Export a few more symbols (#492114).
Stefan Kost [Thu, 1 Nov 2007 08:06:13 +0000 (08:06 +0000)]
gst-libs/gst/audio/audio.*: Readd the deprecation guards, but preserve compilability.
Original commit message from CVS:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/audio.h:
Readd the deprecation guards, but preserve compilability.
Tim-Philipp Müller [Wed, 31 Oct 2007 17:54:48 +0000 (17:54 +0000)]
gst/audioconvert/gstaudioconvert.c: Preserve channel layout when fixating the number of channels in the output caps, ...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (find_suitable_channel_layout),
(gst_audio_convert_fixate_channels), (gst_audio_convert_fixate_caps):
Preserve channel layout when fixating the number of channels in the
output caps, or make sure there's a suitable channel position layout
set on the caps if required. Fixes #430677.
Tim-Philipp Müller [Wed, 31 Oct 2007 17:32:22 +0000 (17:32 +0000)]
tests/check/elements/decodebin.c: Make sure the pipeline really operates in push mode as it should in this case.
Original commit message from CVS:
* tests/check/elements/decodebin.c: (test_text_plain_streams):
Make sure the pipeline really operates in push mode as it should
in this case.
Tim-Philipp Müller [Wed, 31 Oct 2007 15:30:15 +0000 (15:30 +0000)]
gst-libs/gst/audio/audio.h: Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or compilation with DISABLE_...
Original commit message from CVS:
* gst-libs/gst/audio/audio.h:
Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or
compilation with DISABLE_DEPRECATED and without REMOVE_DEPRECATED
(ie. normal cvs builds) will fail.
Stefan Kost [Wed, 31 Oct 2007 12:47:41 +0000 (12:47 +0000)]
tell gtk-doc about the deprecation guard. Apply more doc fixes.
Original commit message from CVS:
* docs/libs/Makefile.am:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/audio.h:
* gst-libs/gst/interfaces/mixer.c:
tell gtk-doc about the deprecation guard. Apply more doc fixes.
Tim-Philipp Müller [Wed, 31 Oct 2007 12:30:28 +0000 (12:30 +0000)]
tests/check/libs/audio.c: Add simple unit test to make sure GstValue intersection of channel layouts works the way I ...
Original commit message from CVS:
* tests/check/libs/audio.c: (init_value_to_channel_layout),
(test_channel_layout_value_intersect), (audio_suite):
Add simple unit test to make sure GstValue intersection
of channel layouts works the way I think it does.
Stefan Kost [Tue, 30 Oct 2007 20:32:14 +0000 (20:32 +0000)]
Fix the docs according to what gtk-doc complained about.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/interfaces/mixer.h:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
* gst-libs/gst/sdp/gstsdpmessage.c:
Fix the docs according to what gtk-doc complained about.
Stefan Kost [Tue, 30 Oct 2007 19:46:02 +0000 (19:46 +0000)]
tests/icles/stress-playbin.c: Fix the build.
Original commit message from CVS:
* tests/icles/stress-playbin.c:
Fix the build.
Tim-Philipp Müller [Tue, 30 Oct 2007 15:54:46 +0000 (15:54 +0000)]
gst/playback/: Post nice/more useful error message if we don't have a decoder for the primary type.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (close_pad_link), (type_found):
* gst/playback/gstdecodebin2.c: (analyze_new_pad):
Post nice/more useful error message if we don't have a decoder for
the primary type.
Wim Taymans [Tue, 30 Oct 2007 15:07:58 +0000 (15:07 +0000)]
gst/playback/gstdecodebin2.c: Be a bit more useful, unblock the pads after we fired the no-more-pads signal so that w...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_group_expose):
Be a bit more useful, unblock the pads after we fired the no-more-pads
signal so that we can use the signal to inspect and connect all pads
without having to keep extra state outside of decodebin.
Wim Taymans [Tue, 30 Oct 2007 15:00:06 +0000 (15:00 +0000)]
gst/playback/gsturidecodebin.c: Implement default signal handler so that we return TRUE when nothing is connected.
Original commit message from CVS:
* gst/playback/gsturidecodebin.c:
(gst_uri_decode_bin_autoplug_continue),
(gst_uri_decode_bin_class_init), (no_more_pads_full):
Implement default signal handler so that we return TRUE when nothing is
connected.
Sebastian Dröge [Sun, 28 Oct 2007 11:53:36 +0000 (11:53 +0000)]
gst-libs/gst/riff/riff-media.c: Use the ALSA channel layout as default for wav files without channel layout informati...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_wavext_add_channel_layout),
(gst_riff_wave_add_default_channel_layout),
(gst_riff_wavext_get_default_channel_mask),
(gst_riff_create_audio_caps):
Use the ALSA channel layout as default for wav files without channel
layout information. This fixes playback of chan-id.wav on 5.1 systems
for example. Also refactor the channel layout setting a bit and add
more default channel orders. Fixes #489010.
Sebastian Dröge [Sun, 28 Oct 2007 11:46:48 +0000 (11:46 +0000)]
Use the ALSA channel layout as default for wav files without channel layout information. This fixes playback of chan-...
Original commit message from CVS:
(gst_riff_wavext_add_channel_layout),
(gst_riff_wave_add_default_channel_layout),
(gst_riff_wavext_get_default_channel_mask),
(gst_riff_create_audio_caps):
Use the ALSA channel layout as default for wav files without channel
layout information. This fixes playback of chan-id.wav on 5.1 systems
for example. Also refactor the channel layout setting a bit and add
more default channel orders. Fixes #489010.
Tim-Philipp Müller [Fri, 26 Oct 2007 18:57:33 +0000 (18:57 +0000)]
tests/check/libs/tag.c: GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
Original commit message from CVS:
* tests/check/libs/tag.c: (test_musicbrainz_tag_registration):
GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
-DGST_DISABLE_DEPRECATED, so use new GST_TAG_ARTIST_SORTNAME
instead.
Christian Schaller [Fri, 26 Oct 2007 12:07:14 +0000 (12:07 +0000)]
update spec file
Original commit message from CVS:
update spec file
Wim Taymans [Thu, 25 Oct 2007 17:36:49 +0000 (17:36 +0000)]
gst/playback/gstdecodebin2.c: Move subtitle encoding property to decodebin2 so that it can set the property value on ...
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
(gst_decode_bin_dispose), (gst_decode_bin_set_caps),
(gst_decode_bin_set_subs_encoding),
(gst_decode_bin_get_subs_encoding), (gst_decode_bin_set_property),
(gst_decode_bin_get_property), (analyze_new_pad):
Move subtitle encoding property to decodebin2 so that it can set the
property value on all elements that it autoplugs and that require it.
Make caps refcounting more consistent in get/set.
* gst/playback/gsturidecodebin.c: (_gst_boolean_accumulator),
(gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
(gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
(gst_uri_decode_bin_get_property), (proxy_unknown_type_signal),
(proxy_autoplug_continue_signal),
(proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
(make_decoder):
Proxy properties and relevant signals from the internal decodebin.
Make properties MT safe.
Tim-Philipp Müller [Thu, 25 Oct 2007 15:10:59 +0000 (15:10 +0000)]
gst-libs/gst/tag/: Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
Original commit message from CVS:
* gst-libs/gst/tag/tag.h: (GST_TAG_MUSICBRAINZ_SORTNAME):
* gst-libs/gst/tag/tags.c:
Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
GST_TAG_ARTIST_SORTNAME (in an API and ABI compatible way).
* gst-libs/gst/tag/gstid3tag.c: (tag_matches):
Map ID3v2 TSOP, TSOA and TSOT frames to new SORTNAME tags (#414539).
* gst-libs/gst/tag/gstvorbistag.c: (tag_matches),
(gst_tag_to_vorbis_comments):
Map new SORTNAME tags (these tags aren't even semi-official, so I'm
just mapping everything I found in the wild) (#414539).
Wim Taymans [Wed, 24 Oct 2007 11:07:57 +0000 (11:07 +0000)]
gst/playback/: Remove the autoplug-sort signal and replace it with a binding friendly autoplug-select signal.
Original commit message from CVS:
Inspired by patch of: René Stadler <mail at renestadler dot de>
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
(gst_decode_bin_autoplug_continue),
(gst_decode_bin_autoplug_factories),
(gst_decode_bin_autoplug_select), (analyze_new_pad), (connect_pad),
(find_compatibles):
* gst/playback/gstplay-marshal.list:
Remove the autoplug-sort signal and replace it with a binding friendly
autoplug-select signal.
Add an autoplug-factories signal that can be used to generate a list of
factories to try to autoplug.
Add the GstPad to the autoplugging signal args as it might be needed to
make a good factory selection.
Fix up the marshallers for this. Fixes #407282.
Tim-Philipp Müller [Tue, 23 Oct 2007 14:23:14 +0000 (14:23 +0000)]
gst-libs/gst/tag/gsttagdemux.c: Don't abort with an assertion if we receive a seek event with a start type of NONE (s...
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c:
Don't abort with an assertion if we receive a seek event with
a start type of NONE (see launchpad bug #155878).
Wim Taymans [Mon, 22 Oct 2007 10:21:46 +0000 (10:21 +0000)]
sys/: Make sure that before we clean up the X resources, we shutdown and join the event thread.
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_event_thread),
(gst_ximagesink_xcontext_get), (gst_ximagesink_xcontext_clear),
(gst_ximagesink_change_state), (gst_ximagesink_reset):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_event_thread),
(gst_xvimagesink_xcontext_get), (gst_xvimagesink_xcontext_clear),
(gst_xvimagesink_change_state), (gst_xvimagesink_reset):
Make sure that before we clean up the X resources, we shutdown and join
the event thread.
Also make sure the event thread does not shut down immediatly after
startup because the running variable is not yet correctly set.
Fixes #378770.
Wim Taymans [Tue, 16 Oct 2007 16:48:38 +0000 (16:48 +0000)]
gst/playback/gstdecodebin.c: Make the window for a race in typefind and shutting down smaller until we figure out the...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (new_pad), (type_found):
Make the window for a race in typefind and shutting down smaller until
we figure out the right locking here. Avoids #485753 usually.
* gst/playback/gstdecodebin2.c: (type_found), (pad_added_group_cb):
Remove unneeded lock causing a race in typefind and shutting down.
Fixes #485753.
* gst/playback/gstplaybin.c: (gst_play_bin_change_state):
Also remove sinks when going to NULL because we might not complete the
state change to PAUSED, causing the PAUSED->READY state change not to
happen.
Wim Taymans [Tue, 16 Oct 2007 15:33:31 +0000 (15:33 +0000)]
gst-libs/gst/audio/gstbaseaudiosink.c: Also explicitly release the ringbuffer when going to NULL because it is requir...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_change_state):
Also explicitly release the ringbuffer when going to NULL because it
is required in the setcaps function, before the state change to PAUSED
completes.
Tim-Philipp Müller [Tue, 16 Oct 2007 14:58:53 +0000 (14:58 +0000)]
tests/icles/: Does what it says on the tin.
Original commit message from CVS:
* tests/icles/.cvsignore:
* tests/icles/Makefile.am:
* tests/icles/stress-playbin.c:
Does what it says on the tin.
Wim Taymans [Mon, 15 Oct 2007 11:38:39 +0000 (11:38 +0000)]
gst/playback/gstqueue2.c: Fix queue negotiation. See #486758.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_push_one):
Fix queue negotiation. See #486758.
Jan Schmidt [Fri, 12 Oct 2007 10:52:18 +0000 (10:52 +0000)]
Actual code change to go along with:
Original commit message from CVS:
Actual code change to go along with:
2007-10-12 Jan Schmidt <Jan.Schmidt@sun.com>
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_decorate),
(gst_xvimagesink_xwindow_new),
(gst_xvimagesink_update_colorbalance),
(gst_xvimagesink_handle_xevents), (gst_xvimagesink_xcontext_get):
Fix handling of some of the X atoms. If the last parameter is True,
XInternAtom won't create the atom if it doesn't exist, and therefore
might return None. This causes X errors on Xv implementations that
don't provide the colour balance attributes.
Jan Schmidt [Fri, 12 Oct 2007 10:37:09 +0000 (10:37 +0000)]
Remove stray character from the changelog.
Original commit message from CVS:
Remove stray character from the changelog.
Jan Schmidt [Fri, 12 Oct 2007 10:33:27 +0000 (10:33 +0000)]
I'm too lazy to comment this
Original commit message from CVS:
*** empty log message ***
Tim-Philipp Müller [Thu, 11 Oct 2007 18:24:09 +0000 (18:24 +0000)]
Extract vorbis comment LICENSE tags correctly.
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c:
* tests/check/libs/tag.c:
Extract vorbis comment LICENSE tags correctly.
Jason Kivlighn [Thu, 11 Oct 2007 16:12:21 +0000 (16:12 +0000)]
Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
Original commit message from CVS:
Patch by: Jason Kivlighn <jkivlighn gmail com>
* gst-libs/gst/tag/gstid3tag.c:
* tests/check/libs/tag.c:
Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
Tim-Philipp Müller [Wed, 10 Oct 2007 17:01:51 +0000 (17:01 +0000)]
gst-libs/gst/tag/gsttagdemux.c: Don't error out when a buggy downstream element doesn't handle the newsegment event w...
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c:
Don't error out when a buggy downstream element doesn't
handle the newsegment event we send properly (especially
not without posting a meaningful error message on the
bus). See bug #471370 and launchpad bug #136264.
Wim Taymans [Wed, 10 Oct 2007 15:36:56 +0000 (15:36 +0000)]
gst-libs/gst/audio/gstbaseaudiosink.c: Use new basesink method to make our EOS drain interruptable.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain):
Use new basesink method to make our EOS drain interruptable.
Jan Schmidt [Wed, 10 Oct 2007 09:37:09 +0000 (09:37 +0000)]
gst-libs/gst/rtp/gstrtppayloads.c: Fix silly search-replace oversight.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtppayloads.c:
Fix silly search-replace oversight.
Laurent Glayal [Tue, 9 Oct 2007 09:57:17 +0000 (09:57 +0000)]
gst-libs/gst/rtp/gstbasertppayload.c: Fix caps memleak. Fixes #484989.
Original commit message from CVS:
Patch by: Laurent Glayal <spglegle at yahoo dot fr>
* gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
(gst_basertppayload_set_outcaps):
Fix caps memleak. Fixes #484989.
Wim Taymans [Mon, 8 Oct 2007 18:04:34 +0000 (18:04 +0000)]
gst-libs/gst/rtp/gstbasertpdepayload.c: Fix debug output.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain):
Fix debug output.
Wim Taymans [Mon, 8 Oct 2007 18:02:53 +0000 (18:02 +0000)]
gst-libs/gst/audio/gstbaseaudiosrc.c: Also handle the case where there is no clock set on the audio source, like in t...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Also handle the case where there is no clock set on the audio source,
like in the unit tests.
Jan Schmidt [Mon, 8 Oct 2007 17:40:17 +0000 (17:40 +0000)]
gst-libs/gst/rtp/gstrtppayloads.c: Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8 to avoid compiler war...
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtppayloads.c:
Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8
to avoid compiler warnings
Wim Taymans [Mon, 8 Oct 2007 17:12:32 +0000 (17:12 +0000)]
gst/playback/: Don't disconnect the have_type signal because we never reconnect it later on. Instead keep a variable ...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (type_found),
(gst_decode_bin_change_state):
* gst/playback/gstdecodebin2.c: (type_found),
(gst_decode_bin_change_state):
Don't disconnect the have_type signal because we never reconnect it
later on. Instead keep a variable to see if we already detected a type.
Wim Taymans [Mon, 8 Oct 2007 10:47:26 +0000 (10:47 +0000)]
gst/playback/: Unlink the signal handler when we found the type, we're not going to do anything sensible with more ty...
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (add_fakesink), (type_found):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
(type_found):
Unlink the signal handler when we found the type, we're not going to do
anything sensible with more type_found signals anyway.
Sebastian Dröge [Mon, 8 Oct 2007 06:07:22 +0000 (06:07 +0000)]
ext/gio/gstgio.c: Use GIO function to get a list of supported URI schemes instead of hard coding something.
Original commit message from CVS:
* ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
Use GIO function to get a list of supported URI schemes instead of
hard coding something.
Tim-Philipp Müller [Sat, 6 Oct 2007 16:49:55 +0000 (16:49 +0000)]
gst-libs/gst/tag/gsttagdemux.c: Don't leak caps.
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c:
Don't leak caps.
Tim-Philipp Müller [Sat, 6 Oct 2007 15:04:53 +0000 (15:04 +0000)]
gst-libs/gst/tag/: API: add GstTagDemux base class for simple tag demuxers.
Original commit message from CVS:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/gsttagdemux.c:
* gst-libs/gst/tag/gsttagdemux.h:
API: add GstTagDemux base class for simple tag demuxers.
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
Add GstTagDemux to docs.
Sebastian Dröge [Fri, 5 Oct 2007 07:49:25 +0000 (07:49 +0000)]
gst-libs/gst/rtp/gstrtpbuffer.c: Fix bug introduced with last commit which inverted the logic and caused all buffers ...
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_get_payload_subbuffer):
Fix bug introduced with last commit which inverted the logic and
caused all buffers to be dropped. Fixes #483620.
Thanks to Laurent Glayal <spglegle at yahoo dot fr> for noticing.