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.
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.
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.
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.
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
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.
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.
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()
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.
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.
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.
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.
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.
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.
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.
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).
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"
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
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
Original commit message from CVS:
* docs/design/design-decodebin.txt:
* gst/playback/gstdecodebin2.c: (analyze_new_pad):
Update some more docs and comments.
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).
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.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (queue_threshold_reached),
(finish_source):
Avoid crash when there are external subtitles (fixes#491722).
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.
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).
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.
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).
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).
Original commit message from CVS:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/audio.h:
Readd the deprecation guards, but preserve compilability.
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.
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.
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.
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.
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.
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.
