ChangeLog 374 KB
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2007-02-14  Wim Taymans,,,  <wim@fluendo.com>

	Patch by: jp.liu <jp_liu at astrocom dot cn>

	* gst/rtsp/rtspurl.c: (rtsp_url_parse):
	Fix parsing of password field in url. Fixes #407797.

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2007-02-14  Wim Taymans,,,  <wim@fluendo.com>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_class_init),
	(gst_wavparse_reset), (gst_wavparse_init),
	(gst_wavparse_destroy_sourcepad), (gst_wavparse_fmt),
	(gst_wavparse_parse_file_header), (gst_wavparse_stream_init),
	(gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info),
	(gst_wavparse_stream_headers), (gst_wavparse_parse_stream_init),
	(gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
	(gst_wavparse_loop), (gst_wavparse_chain),
	(gst_wavparse_pad_convert), (gst_wavparse_pad_query),
	(gst_wavparse_srcpad_event), (gst_wavparse_change_state),
	(plugin_init):
	* gst/wavparse/gstwavparse.h:
	Update docs.
	Use boilerplate.
	Various code cleanups.
	When the bitrate is not known (bps == 0 or compressed formats) let
	downstream element guestimate the duration and position and don't
	generate timestamps or durations. Fixes #405213.
	Fix EOS and ERROR conditions in chain mode, we just need to forward the
	error flowreturn upstream.

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2007-02-13  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/gconf/Makefile.am:
	* ext/gconf/gconf.c: (gst_gconf_get_string),
	(gst_gconf_get_key_for_sink_profile), (gst_gconf_set_string),
	(gst_gconf_render_bin_with_default):
	* ext/gconf/gconf.h:
	* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init),
	(gst_gconf_audio_sink_reset), (gst_gconf_audio_sink_init),
	(gst_gconf_audio_sink_dispose), (do_change_child),
	(gst_gconf_switch_profile), (gst_gconf_audio_sink_set_property),
	(cb_change_child), (gst_gconf_audio_sink_change_state):
	* ext/gconf/gstgconfaudiosink.h:
	* ext/gconf/gstswitchsink.c: (gst_switch_sink_base_init),
	(gst_switch_sink_class_init), (gst_switch_sink_reset),
	(gst_switch_sink_init), (gst_switch_sink_dispose),
	(gst_switch_commit_new_kid), (gst_switch_sink_set_child),
	(gst_switch_sink_set_property), (gst_switch_sink_handle_event),
	(gst_switch_sink_get_property), (gst_switch_sink_change_state):
	* ext/gconf/gstswitchsink.h:
	* gst/autodetect/gstautoaudiosink.c:
	(gst_auto_audio_sink_class_init), (gst_auto_audio_sink_dispose),
	(gst_auto_audio_sink_clear_kid), (gst_auto_audio_sink_reset),
	(gst_auto_audio_sink_detect):
	* gst/autodetect/gstautovideosink.c:
	(gst_auto_video_sink_class_init), (gst_auto_video_sink_dispose),
	(gst_auto_video_sink_clear_kid), (gst_auto_video_sink_reset),
	(gst_auto_video_sink_detect):
	Re-factor the gconfaudiosink into a "GstSwitchSink" base class
	and a child that implements the GConf key monitoring. The end goal of
	this is an audio sink that can be changed on the fly, but at the 
	moment it still only changes on the next READY transition.

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2007-02-13  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
	(gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
	(gst_avi_demux_sync), (gst_avi_demux_massage_index),
	(gst_avi_demux_calculate_durations_from_index),
	(gst_avi_demux_push_event), (gst_avi_demux_stream_header_pull),
	(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
	(gst_avi_demux_loop):
	  Put debug stuff into #ifndef GST_DISABLE_DEBUG #endif

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2007-02-13  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	* docs/plugins/Makefile.am:
	  Add crossreferences to glib/gobject/gstream docs.

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2007-02-12  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/monoscope/Makefile.am:
	* gst/monoscope/gstmonoscope.c:
	  Fix copy'n'paste-o in docs chunk. Also add some missing CFLAGS
	  (but no LIBS, since we only use defines from the headers).

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2007-02-12  Tim-Philipp Müller  <tim at centricular dot net>

	Based on patch by: Jonathan Matthew  <jonathan at kaolin wh9 net>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_parse_stream_init),
	(gst_wavparse_stream_data):
	  Fix massive memory leak when operating in streaming mode due to
	  GST_BUFFER_MALLOCDATA() not being set on newly-created buffers.
	  Fixes #407057.

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2007-02-12  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
	(gst_avi_demux_reset), (gst_avi_demux_index_entry_for_time),
	(gst_avi_demux_handle_src_query), (gst_avi_demux_parse_superindex),
	(gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream),
	(gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
	(gst_avi_demux_sync), (gst_avi_demux_next_data_buffer),
	(gst_avi_demux_stream_scan), (gst_avi_demux_massage_index),
	(gst_avi_demux_calculate_durations_from_index),
	(gst_avi_demux_push_event), (gst_avi_demux_stream_header_pull),
	(gst_avi_demux_do_seek), (gst_avi_demux_process_next_entry),
	(gst_avi_demux_stream_data), (gst_avi_demux_loop):
	* gst/avi/gstavidemux.h:
	  Save some memory (8%) by repacking the index entry structure (more to
	  come). Add more FIXMEs to questionable parts.

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2007-02-12  Stefan Kost  <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2src.c: (gst_v4l2src_v4l2fourcc_to_caps),
	(gst_v4l2src_get_caps):
	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
	(gst_v4l2src_grab_frame), (gst_v4l2src_set_capture),
	(gst_v4l2src_capture_init):
	  More FIXME comments and messaging changes.

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2007-02-12  Stefan Kost  <ensonic@users.sf.net>

	* gst/goom/gstgoom.c: (gst_goom_class_init), (gst_goom_init),
	(gst_goom_change_state):
	* gst/goom/gstgoom.h:
	  Improved docs and use GST_DEBUG_FUNCPTR.

	* gst/level/gstlevel.c: (gst_level_class_init):
	  Use GST_DEBUG_FUNCPTR.

	* gst/monoscope/gstmonoscope.c: (gst_monoscope_init),
	(gst_monoscope_chain), (gst_monoscope_change_state):
	  Improved docs source cleanups.

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2007-02-12  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/debug/Makefile.am:
	* gst/debug/gstdebug.c: (plugin_init):
	* gst/debug/gstpushfilesrc.c:
	* gst/debug/gstpushfilesrc.h:
	  Add code for a pushfilesrc element that implements a pushfile:// URI
	  handler, to make debugging push-mode operation of demuxer/decoders
	  that support both easier in connection with seek/playbin/etc.
	  The element isn't registered at the moment.

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2007-02-11  Sébastien Moutte  <sebastien@moutte.net>

	* gst/avi/gstavimux.c:
	  Comment a #if 0 in caps template definition as VS6 seems to 
	do not support it.
	* gst/rtsp/gstrtspsrc.c:(gst_rtspsrc_loop_udp):
	  Use gst_guint64_to_gdouble for conversion.
	* gst/rtsp/rtspconnection.c:(rtsp_connection_send):
	  Move variables declaration before the first instruction.
	* gst/rtsp/rtspdefs.c:(rtsp_strresult):
	  Don't use hstrerror for error log on G_OS_WIN32 build as it's not supported.
	  And don't include netdb.h for G_OS_WIN32
	* gst/rtsp/sdpmessage.c:(sdp_parse_line):
	  This initialization SDPMedia nmedia = {.media = NULL }; is not supported
	  by VS6 then use an other way to initialize SDPMedia structure.
	* gst/udp/gstdynudpsink.h:
	* gst/udp/gstdynudpnetutils.h:
	  Do not include <sys/time.h> for G_OS_WIN32
	* gst/udp/gstudpsrc.c:
	  Define socklen_t as int for G_OS_WIN32
	* win/common/config.h.in:
	  Undef HAVE_NETINET_IN_H
	* win32/vs6/gst_plugins_good.dsw:
	* win32/vs6/libgstrtp.dsp:
	* win32/vs6/libgstrtsp.dsp:
	* win32/vs6/libgstautogen.dsp:
	* win32/vs6/libgstaudiofx.dsp:
	* win32/vs6/libgstudp.dsp:
	  Add and update project files.
	* win32/common/gstudp-enumtypes.c:
	* win32/common/gstudp-enumtypes.h:
	  Add a copy of udp enumtypes to win32/common as in core 
	  and base.
	
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2007-02-11  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	  Activate monoscope when building with --enable-experimental. Fix
	  --enable-external configure switch description.

	* sys/sunaudio/gstsunaudiomixer.c: (gst_sunaudiomixer_base_init):
	* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_dispose):
	  Help gst-indent.

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2007-02-09  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/avi/gstavimux.c: (gst_avi_mux_riff_get_avi_header):
	  Explicitly cast result of pointer arithmetic to integer in order to
	  avoid compiler warnings on some 64-bit systems. Should fix #406018.

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2007-02-08  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/debug/progressreport.c:
	  Some more docs.

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2007-02-07  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/plugins/inspect/plugin-rtp.xml:
	  Update for new elements.

	* gst/debug/progressreport.h:
	  Commit newly-created header file as well.

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2007-02-07  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* gst/debug/Makefile.am:
	* gst/debug/progressreport.c: (gst_progress_report_post_progress),
	(gst_progress_report_do_query), (gst_progress_report_report):
	  Make progressreport element post messages with the current progress
	  on the bus. Also add some basic docs for it.

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2007-01-30  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/hal/hal.c: (gst_hal_get_string):
	* ext/hal/hal.h:
	  Some small cleanups; deal with errors when parsing the HAL ALSA
	  capabilities a bit better.

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2007-02-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/smpte/gstsmpte.c: (gst_smpte_transition_type_get_type):
	  Let's try this again and use the right cast this time.

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2007-02-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/smpte/gstsmpte.c: (gst_smpte_transition_type_get_type):
	  Add cast to avoid compiler warnings with older GLib versions
	  where the nick/name members in GEnumValue are not declared as
	  constant strings.

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2007-02-06  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/gconf/gconf.c: (gst_gconf_get_key_for_sink_profile),
	(gst_gconf_render_bin_from_key),
	(gst_gconf_get_default_audio_sink):
	* ext/gconf/gconf.h:
	* ext/gconf/gstgconfaudiosink.c: (get_gconf_key_for_profile),
	(do_toggle_element), (gst_gconf_audio_sink_set_property),
	(gst_gconf_audio_sink_get_property):
	  In gconfaudiosink, get the right key as the old key in do_toggle
	  (ie. one dependent on the profile selected). Log some more stuff so
	  we can see what's actually going on.

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2007-02-06  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audiofx/audioamplify.c: (gst_audio_amplify_base_init),
	(gst_audio_amplify_class_init), (gst_audio_amplify_init),
	(gst_audio_amplify_set_process_function),
	(gst_audio_amplify_setup):
	* gst/audiofx/audioamplify.h:
	* gst/audiofx/audioinvert.c: (gst_audio_invert_base_init),
	(gst_audio_invert_class_init), (gst_audio_invert_setup):
	* gst/audiofx/audioinvert.h:
	Some small cleanups and port both elements to the new GstAudioFilter
	base class to save a few lines of common code.
	* gst/audiofx/Makefile.am:
	Link against libgstaudio for the above changes

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2007-01-29  Wim Taymans  <wim@fluendo.com>

	* tests/check/elements/.cvsignore:
	Some more ignores.

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2007-01-26  Wim Taymans  <wim@fluendo.com>

	Patch by: charles <charlesg3 at gmail dot com>

	* ext/shout2/gstshout2.c: (gst_shout2send_init),
	(set_shout_metadata), (gst_shout2send_event):
	* ext/shout2/gstshout2.h:
	Properly handle tags in shout2send. Fixes #399825.

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2007-01-25  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_activate_streams):
	Convert SDP fields to upper/lowercase following the rules in the SDP to
	caps document. 

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2007-01-25  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/README:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpilbcpay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbispay.c:
	Fix case of encoding-name and key/value pairs to match the document.
	This is to make interoperation with SDP case-insensitive as required by
	the relevant RFCs.

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	* configure.ac:
	Bump required -core/-base to CVS

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	* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_setcaps),
	(gst_rtp_L16_pay_flush), (gst_rtp_L16_pay_handle_buffer):
	* gst/rtp/gstrtpL16pay.h:
	Fill up to MTU using adapter.
	Timestamp rtp packets.

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	* gst/multipart/multipartmux.c: (gst_multipart_mux_collected):
	* sys/ximage/ximageutil.c: (ximageutil_check_xshm_calls):
	Use G_GSIZE_FORMAT in print statements for portability.
	Fixes build on macosx.

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2007-01-24  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_base_init),
	(gst_rtp_L16_depay_class_init), (gst_rtp_L16_depay_init),
	(gst_rtp_L16_depay_parse_int), (gst_rtp_L16_depay_setcaps),
	(gst_rtp_L16_depay_process), (gst_rtp_L16_depay_set_property),
	(gst_rtp_L16_depay_get_property), (gst_rtp_L16_depay_change_state),
	(gst_rtp_L16_depay_plugin_init):
	* gst/rtp/gstrtpL16depay.h:
	* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_get_type),
	(gst_rtp_L16_pay_base_init), (gst_rtp_L16_pay_class_init),
	(gst_rtp_L16_pay_init), (gst_rtp_L16_pay_finalize),
	(gst_rtp_L16_pay_setcaps), (gst_rtp_L16_pay_handle_buffer),
	(gst_rtp_L16_pay_plugin_init):
	* gst/rtp/gstrtpL16pay.h:
	Port and enable raw audio payloader/depayloader. Needs a bit more work
	on the payloader side.

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2007-01-24  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (pad_blocked),
	(gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_udp):
	* gst/rtsp/gstrtspsrc.h:
	Only unblock the udp pads when we linked and activated them all.
	Fixes #395688.

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2007-01-24  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpac3depay.c: (gst_rtp_ac3_depay_base_init),
	(gst_rtp_ac3_depay_class_init), (gst_rtp_ac3_depay_init),
	(gst_rtp_ac3_depay_setcaps), (gst_rtp_ac3_depay_process),
	(gst_rtp_ac3_depay_set_property), (gst_rtp_ac3_depay_get_property),
	(gst_rtp_ac3_depay_change_state), (gst_rtp_ac3_depay_plugin_init):
	* gst/rtp/gstrtpac3depay.h:
	Added simple AC3 depayloader (RFC 4184).

	* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps):
	Fix a leak.

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2007-01-24  Sebastian Dröge  <slomo@circular-chaos.org>

	reviewed by: Stefan Kost  <ensonic@users.sf.net>

	* gst/audiofx/Makefile.am:
	* gst/audiofx/audioamplify.c:
	(gst_audio_amplify_clipping_method_get_type),
	(gst_audio_amplify_base_init), (gst_audio_amplify_class_init),
	(gst_audio_amplify_init), (gst_audio_amplify_set_process_function),
	(gst_audio_amplify_set_property), (gst_audio_amplify_get_property),
	(gst_audio_amplify_set_caps),
	(gst_audio_amplify_transform_int_clip),
	(gst_audio_amplify_transform_int_wrap_negative),
	(gst_audio_amplify_transform_int_wrap_positive),
	(gst_audio_amplify_transform_float_clip),
	(gst_audio_amplify_transform_float_wrap_negative),
	(gst_audio_amplify_transform_float_wrap_positive),
	(gst_audio_amplify_transform_ip):
	* gst/audiofx/audioamplify.h:
	* gst/audiofx/audiofx.c: (plugin_init):
	Add new element "audioamplify". This allows scaling of raw audio
	samples, similar to the "volume" element, but provides different modes
	for clipping and allows unlimited amplification. It's mainly targeted
	for creative sound design and not as a replacement of the "volume"
	element. Fixes #397162
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-audiofx.xml:
	Add docs for audioamplify and integrate them into the build system
	* tests/check/Makefile.am:
	* tests/check/elements/audioamplify.c: (setup_amplify),
	(cleanup_amplify), (GST_START_TEST), (amplify_suite), (main):
	Add fairly extensive unit test suite for audioamplify

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2007-01-24  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (pad_unblocked), (pad_blocked):
	Unblock pads after adding the pads to the element so that autopluggers
	get a change to link something. Possibly fixes #395688.

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2007-01-24  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtpilbcdepay.c:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init):
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_class_init),
	(gst_rtp_mpa_depay_init), (gst_rtp_mpa_depay_setcaps),
	(gst_rtp_mpa_depay_process):
	* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_class_init),
	(gst_rtp_mpv_depay_init), (gst_rtp_mpv_depay_process):
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbispay.c:
	Fix caps with payload numbers.
	Add some fixed payload numbers to caps when possible.

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2007-01-23  Sebastian Dröge  <slomo@circular-chaos.org>

	reviewed by: Stefan Kost  <ensonic@users.sf.net>

	* gst/audiofx/Makefile.am:
	* gst/audiofx/audiofx.c: (plugin_init):
	* gst/audiofx/audioinvert.c: (gst_audio_invert_base_init),
	(gst_audio_invert_class_init), (gst_audio_invert_init),
	(gst_audio_invert_set_property), (gst_audio_invert_get_property),
	(gst_audio_invert_set_caps), (gst_audio_invert_transform_int),
	(gst_audio_invert_transform_float),
	(gst_audio_invert_transform_ip):
	* gst/audiofx/audioinvert.h:
	Add new audiofx element "audioinvert". This element swaps the upper
	and lower half of samples and can be used for example for a
	wide-stereo effect. Fixes #396057
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/inspect/plugin-audiofx.xml:
	Add docs for the audioinvert element and add them to the build system.
	* tests/check/Makefile.am:
	* tests/check/elements/audioinvert.c: (setup_invert),
	(cleanup_invert), (GST_START_TEST), (invert_suite), (main):
	Add unit test suite for the audioinvert element.

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2007-01-23  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_parse_int),
	(gst_rtp_mp4g_depay_setcaps), (gst_rtp_mp4g_depay_process):
	Parse config params as string and int.
	Parse and use AU header length

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2007-01-23  Wim Taymans  <wim@fluendo.com>

	* gst/smpte/barboxwipes.c: (gst_wipe_boxes_draw),
	(gst_wipe_triangles_clock_draw), (gst_wipe_triangles_draw):
	* gst/smpte/gstmask.c: (_gst_mask_register):
	* gst/smpte/gstmask.h:
	* gst/smpte/gstsmpte.c: (gst_smpte_update_mask):
	* gst/smpte/paint.c: (gst_smpte_paint_hbox), (draw_bresenham_line),
	(gst_smpte_paint_triangle_clock):
	constify some static structs.
	Don't update the mask if nothing changed to the params.
	Make sure we never draw outside of the picture. Fixes #398325.

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2007-01-22  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_stream_header_pull):
	  Error out properly when pull_range fails while we're reading the
	  headers, instead of just pausing the task silently. Fixes #399338.

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2007-01-19  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/smpte/gstsmpte.c: (gst_smpte_collected):
	  Some more sanity checks to make sure the input formats match and the
	  input pads are actually negotiated, in case someone tries to feed
	  buffers from fakesrc or filesrc. Fixes #398299.
	  Also const-ify an array, just because we can.

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	* gst/smpte/gstsmpte.c: (fill_i420), (gst_smpte_collected):
	Ignore previous commit, that was only valid for widths and heights
	that are multiples of 4.
	Copy over size/stride macros from jpegdec. This allows the element
	to work with any width,height...
	... but puts in evidence that the actual transformations only work
	with width/height that are multiples of 4.

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2007-01-19  Edward Hervey  <edward@fluendo.com>

	* gst/smpte/gstsmpte.c: (gst_smpte_collected):
	Allocate buffers of the right size.
	The proper size of a I420 buffer in bytes is:
	
	    width * height * 3
	    ------------------
	            2

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2007-01-18  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/smpte/gstsmpte.c: (gst_smpte_init):
	  Proxy getcaps on sink pads too, so that we either end up with the
	  same dimensions on all pads or error out if that's not possible
	  (seems to work even!). Fixes #398086, I think.

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2007-01-18  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	  Remove ladspa from docs; add hierarchy info for GstAudioPanorama;
	  fix integer properties with -1 as minimum value.

	* docs/plugins/inspect/plugin-1394.xml:
	* docs/plugins/inspect/plugin-aasink.xml:
	* docs/plugins/inspect/plugin-alaw.xml:
	* docs/plugins/inspect/plugin-alpha.xml:
	* docs/plugins/inspect/plugin-alphacolor.xml:
	* docs/plugins/inspect/plugin-annodex.xml:
	* docs/plugins/inspect/plugin-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-auparse.xml:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-avi.xml:
	* docs/plugins/inspect/plugin-cacasink.xml:
	* docs/plugins/inspect/plugin-cairo.xml:
	* docs/plugins/inspect/plugin-cdio.xml:
	* docs/plugins/inspect/plugin-cutter.xml:
	* docs/plugins/inspect/plugin-debug.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-efence.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-esdsink.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-halelements.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.xml:
	* docs/plugins/inspect/plugin-mulaw.xml:
	* docs/plugins/inspect/plugin-multipart.xml:
	* docs/plugins/inspect/plugin-navigationtest.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-smpte.xml:
	* docs/plugins/inspect/plugin-speex.xml:
	* docs/plugins/inspect/plugin-taglib.xml:
	* docs/plugins/inspect/plugin-udp.xml:
	* docs/plugins/inspect/plugin-videobalance.xml:
	* docs/plugins/inspect/plugin-videobox.xml:
	* docs/plugins/inspect/plugin-videoflip.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-wavenc.xml:
	* docs/plugins/inspect/plugin-wavparse.xml:
	* docs/plugins/inspect/plugin-ximagesrc.xml:
	  Update to CVS.

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	* gst/audiofx/audiopanorama.c:
	  Fix doc section name (Fixes #397946)

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2007-01-17  Stefan Kost  <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2object.c:
	(gst_v4l2_object_install_properties_helper),
	(gst_v4l2_object_set_property_helper),
	(gst_v4l2_object_get_property_helper), (gst_v4l2_set_defaults):
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init),
	(gst_v4l2src_init), (gst_v4l2src_set_property),
	(gst_v4l2src_get_property), (gst_v4l2src_set_caps):
	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
	(gst_v4l2src_grab_frame), (gst_v4l2src_set_capture),
	(gst_v4l2src_capture_init), (gst_v4l2src_capture_start),
	(gst_v4l2src_capture_deinit):
	  Fix EIO handing when capturing. Add new property to specify the number of
	  buffers to enque (and remove the borked num-buffers usage).

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2007-01-16  Stefan Kost  <ensonic@users.sf.net>

	Patch by: Sebastian Dröge <slomo circular-chaos org>

	* gst/audiofx/audiopanorama.c: (gst_audio_panorama_class_init),
	(gst_audio_panorama_set_process_function):
	  Use a function array for process methods, add more docs and define the
	  startindex of enums.

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2007-01-14  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Mark Nauwelaerts <manauw at skynet be>

	* gst/avi/gstavimux.c: (gst_avi_mux_finalize),
	(gst_avi_mux_pad_reset), (gst_avi_mux_reset), (gst_avi_mux_init),
	(gst_avi_mux_vidsink_set_caps), (gst_avi_mux_audsink_set_caps),
	(gst_avi_mux_request_new_pad), (gst_avi_mux_release_pad),
	(gst_avi_mux_riff_get_avi_header),
	(gst_avi_mux_riff_get_avix_header), (gst_avi_mux_riff_get_header),
	(gst_avi_mux_write_avix_index), (gst_avi_mux_add_index),
	(gst_avi_mux_bigfile), (gst_avi_mux_start_file),
	(gst_avi_mux_stop_file), (gst_avi_mux_handle_event),
	(gst_avi_mux_do_buffer), (gst_avi_mux_do_one_buffer),
	(gst_avi_mux_change_state):
	* gst/avi/gstavimux.h:
	* tests/check/elements/avimux.c: (teardown_src_pad):
	  Add support for more than one audio stream; write better AVIX
	  header; refactor code a bit; don't announce vorbis caps on our audio
	  sink pads since we don't support it anyway. Closes #379298.

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2007-01-13  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Sebastian Dröge <slomo circular-chaos org>

	* gst/audiofx/audiopanorama.c:
	(gst_audio_panorama_method_get_type),
	(gst_audio_panorama_class_init), (gst_audio_panorama_init),
	(gst_audio_panorama_set_process_function),
	(gst_audio_panorama_set_property),
	(gst_audio_panorama_get_property), (gst_audio_panorama_set_caps),
	(gst_audio_panorama_transform_m2s_int_simple),
	(gst_audio_panorama_transform_s2s_int_simple),
	(gst_audio_panorama_transform_m2s_float_simple),
	(gst_audio_panorama_transform_s2s_float_simple):
	* gst/audiofx/audiopanorama.h:
	  Add 'method' property and provide a simple (non-psychoacustic)
	  processing method (#394859).

	* tests/check/elements/audiopanorama.c: (GST_START_TEST),
	(panorama_suite):
	  Tests for new method.

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2007-01-11  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/apetag/gsttagdemux.c: (gst_tag_demux_read_range):
	* gst/id3demux/gstid3demux.c: (gst_id3demux_read_range):
	  Set correct caps on outgoing pulled buffers, or things blow up
	  after recent core changes.

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2007-01-11  Wim Taymans  <wim@fluendo.com>

	Based on patch by: Jonas Holmberg <jonas dot holmberg at axis dot com>

	* gst/multipart/multipartmux.c: (gst_multipart_mux_init),
	(gst_multipart_mux_request_new_pad),
	(gst_multipart_mux_queue_pads), (gst_multipart_mux_collected),
	(gst_multipart_mux_change_state):
	Return FLOW errors ASAP. Fixes #394977.
	Misc cleanups.

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2007-01-11  Wim Taymans  <wim@fluendo.com>

	Patch by: Lutz Mueller <lutz at topfrose dot de>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams):
	Check for stream pad before activating. 

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2007-01-10  Wim Taymans  <wim@fluendo.com>

	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/rtsp/COPYING.MIT:
	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
	(gst_rtspsrc_stream_free), (gst_rtspsrc_cleanup),
	(gst_rtspsrc_alloc_udp_ports), (pad_unblocked), (pad_blocked),
	(gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_send),
	(gst_rtspsrc_parse_methods),
	(gst_rtspsrc_create_transports_string),
	(gst_rtspsrc_prepare_transports), (gst_rtspsrc_setup_streams),
	(gst_rtspsrc_open), (gst_rtspsrc_close):
	* gst/rtsp/gstrtspsrc.h:
	* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	(rtsp_connection_connect), (rtsp_connection_send), (read_line),
	(parse_request_line), (parse_line), (rtsp_connection_read),
	(rtsp_connection_close):
	* gst/rtsp/rtspdefs.c: (rtsp_init_status), (rtsp_strresult),
	(rtsp_method_as_text), (rtsp_header_as_text),
	(rtsp_status_as_text), (rtsp_find_header_field),
	(rtsp_find_method):
	* gst/rtsp/rtspdefs.h:
	* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send),
	(rtsp_ext_wms_configure_stream):
	* gst/rtsp/rtspmessage.c: (rtsp_message_new), (rtsp_message_init),
	(rtsp_message_new_request), (rtsp_message_init_request),
	(rtsp_message_new_response), (rtsp_message_init_response),
	(rtsp_message_init_data), (rtsp_message_unset),
	(rtsp_message_free), (rtsp_message_add_header),
	(rtsp_message_get_header), (rtsp_message_set_body),
	(rtsp_message_get_body), (dump_mem), (rtsp_message_dump):
	* gst/rtsp/rtspmessage.h:
	* gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n),
	(sdp_media_get_attribute_val_n), (read_string), (read_string_del),
	(sdp_parse_line), (sdp_message_parse_buffer), (print_media),
	(sdp_message_dump):
	Allow url to be NULL to be able to use it for server connections.
	Can now send responses as well as requests.
	No longer hangs in an endless loop if EOF is received.
	Can now convert a status code to a text string.
	Return RTSP_HDR_INVALID for unknown headers.
	Return RTSP_INVALID for unknown methods.
	Copy CSeq and Session headers from the request.
	Only free memory corresponding to the currently set message type.
	Added const to function arguments as appropriate.
	Avoid a compiler warning when initializing nmedia.
	Use guint rather than gint to avoid compiler warnings.
	Fix crasher in wms extension.
	Factor out stream setup from open_connection.
	Delay activation of streams when actual data is received from the
	server, this prepares us to do proper protocol switching.
	Added new license.
	Fixes #380895.


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2007-01-10  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Sebastian Dröge <slomo ubuntu com>

	* docs/plugins/Makefile.am:
	* gst/audiofx/audiopanorama.c:
	  Some small docs fixes (#394851).

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2007-01-09  Wim Taymans  <wim@fluendo.com>

	* gst/avi/gstavidemux.c:
	Fix docs.

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2007-01-09  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_base_init),
	(gst_rtp_mpv_depay_class_init), (gst_rtp_mpv_depay_init),
	(gst_rtp_mpv_depay_setcaps), (gst_rtp_mpv_depay_process),
	(gst_rtp_mpv_depay_set_property), (gst_rtp_mpv_depay_get_property),
	(gst_rtp_mpv_depay_change_state), (gst_rtp_mpv_depay_plugin_init):
	* gst/rtp/gstrtpmpvdepay.h:
	  Added RFC 2250 MPEG Video Depayloader.

	* gst/rtp/gstrtpL16depay.h:
	* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps),
	(gst_rtp_h263p_depay_process):
	Fix Header file. Small cleanups.

	* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_class_init),
	(gst_rtp_mp4g_depay_init), (gst_rtp_mp4g_depay_finalize),
	(gst_rtp_mp4g_depay_process), (gst_rtp_mp4g_depay_change_state):
	* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init),
	(gst_rtp_mp4v_depay_init), (gst_rtp_mp4v_depay_finalize),
	(gst_rtp_mp4v_depay_setcaps), (gst_rtp_mp4v_depay_process),
	(gst_rtp_mp4v_depay_change_state):
	Remove usused code. Remove Adapter from state Change. Added debug.

	* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_base_init),
	(gst_rtp_mpa_depay_class_init), (gst_rtp_mpa_depay_init),
	(gst_rtp_mpa_depay_setcaps), (gst_rtp_mpa_depay_process):
	* gst/rtp/gstrtpmpadepay.h:
	Subclass base depayloader.
	Added debug.
	Support static payload type assignment as well.

	* gst/rtp/gstrtpmpapay.c:
	Fix caps.

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2007-01-08  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Vincent Torri  <vtorri at univ-evry fr>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/jpeg/smokecodec.c:
	  These libjpeg callbacks should return a 'boolean' (unsigned char
	  apparently) and not a 'gboolean' (which maps to gint). Fixes
	  warnings when compiling with MingW (#393427).

	* gst/rtsp/rtspconnection.c: (rtsp_connection_read):
	  Use ioctlsocket on win32.

	* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
	  Some printf format fixes for win32.

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2007-01-07  Sébastien Moutte  <sebastien@moutte.net>

	* gst/cutter/gstcutter.c: (gst_cutter_chain):
	  Use gst_guint64_to_gdouble for conversion.
	* win32/vs6/libgstmatroska.dsp:
	  Add zlib to the link.
	* win32/vs6/libgstvideobox.dsp:
	  Update liboil library name (project is linked to 
	  liboil-0.3-0.lib now).
	  
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2007-01-05  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/matroska/Makefile.am:
	  If zlib is available and used, we must link it explicitly for
	  things to work on MingW (fixes #392855).

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2007-01-04  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/esd/esdsink.c: (gst_esdsink_delay):
	  Don't return bogus values when esd_get_delay() fails for some
	  reason (#392189).

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2006-12-24  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/ximage/gstximagesrc.c: (composite_pixel):
	  Fix presumably copy'n'pasto for 16bpp depth.

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2006-12-24  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/matroska/matroska-mux.c:
	(gst_matroska_mux_audio_pad_setcaps):
	  The "signed" field in audio caps is of boolean type, trying to use
	  gst_structure_get_int() to extract it will fail. Fixing this makes
	  matroskamux accept raw audio input (#387121) (use at your own risk
	  though, due to the matroska spec being not entirely useful in this
	  respect).
	  Also fix up raw audio structures in template caps so that they
	  represent what our setcaps function will actually accept, so that
	  converters know what to convert to.
	  Finally, don't fail if there isn't an "endianness" field in 8-bit
	  PCM caps.

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2006-12-22  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/elements/audiopanorama.c: (cleanup_panorama):
	* tests/check/elements/avimux.c: (setup_avimux), (cleanup_avimux):
	* tests/check/elements/cmmldec.c: (setup_cmmldec),
	(teardown_cmmldec):
	* tests/check/elements/cmmlenc.c: (setup_cmmlenc),
	(teardown_cmmlenc):
	* tests/check/elements/level.c: (setup_level), (cleanup_level):
	  reapply consistent pad (de)activation

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2006-12-21  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	Back to CVS

	* gst-plugins-good.doap:
	Add 0.10.5 doap entry

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=== release 0.10.5 ===

2006-12-21  Jan Schmidt <thaytan@mad.scientist.com>

	* configure.ac:
	  releasing 0.10.5, "The Path of Thorns"

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2006-12-21  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/elements/audiopanorama.c: (cleanup_panorama):
	* tests/check/elements/avimux.c: (setup_avimux), (cleanup_avimux):
	* tests/check/elements/cmmldec.c: (setup_cmmldec),
	(teardown_cmmldec):
	* tests/check/elements/cmmlenc.c: (setup_cmmlenc),
	(teardown_cmmlenc):
	* tests/check/elements/level.c: (setup_level), (cleanup_level):
	  revert my freeze breakage

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2006-12-21  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/elements/audiopanorama.c: (cleanup_panorama):
	* tests/check/elements/avimux.c: (setup_avimux), (cleanup_avimux):
	* tests/check/elements/cmmldec.c: (setup_cmmldec),
	(teardown_cmmldec):
	* tests/check/elements/cmmlenc.c: (setup_cmmlenc),
	(teardown_cmmlenc):
	* tests/check/elements/level.c: (setup_level), (cleanup_level):
	  consistent pad (de)activation

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2006-12-18  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	* ext/Makefile.am:
	Disable LADPSA, as it has moved to the -bad module for the duration.

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2006-12-18  Wim Taymans  <wim@fluendo.com>

	* ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_setcaps),
	(gst_signal_processor_event):
	Reset flow_state back to _OK after a flush stop so that we exit our
	error state after the flush. Fixes #374213

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	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  Decent effort at porting to 0.10.  Needs cleanup on OS/X.

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2006-12-16  David Schleef  <ds@schleef.org>

	Patch by: Vijay Santhanam <vijay santhanam gmail com>

	* sys/osxvideo/Makefile.am:
	* sys/osxvideo/osxvideosink.h:
	* sys/osxvideo/osxvideosink.m:
	  Preliminary patch for porting osxvideosink

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2006-12-16  Wim Taymans  <wim@fluendo.com>

	Patch by: Sjoerd Simons <sjoerd at luon dot net>

	* gst/videomixer/videomixer.c: (gst_videomixer_pad_set_property),
	(gst_videomixer_set_master_geometry),
	(gst_videomixer_pad_sink_setcaps), (gst_videomixer_collect_free),
	(gst_videomixer_reset), (gst_videomixer_init),
	(gst_videomixer_finalize), (gst_videomixer_request_new_pad),
	(gst_videomixer_release_pad), (gst_videomixer_collected),
	(gst_videomixer_change_state):
	Introduce some locking around the videomixer state so that it does not
	crash when adding/removing pads. Fixes #383043.

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2006-12-16  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  Make sure libcaca can actually be used instead of just checking for
	  /usr/bin/caca-config, so we don't wrongly try to build cacasink when
	  cross-compiling (fixes #384587).

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2006-12-15  Thomas Vander Stichele  <thomas at apestaart dot org>

	* Makefile.am:
	* gst-plugins-good.doap:
	* gst-plugins-good.spec.in:
	  adding doap file

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2006-12-14  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  libflac-1.1.3 changed API again, but we can't build against it yet,
	  so make sure our check doesn't use libflac-1.1.3 and add a comment
	  to this effect.

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2006-12-14  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/effectv/gstquark.c: (gst_quarktv_transform),
	(gst_quarktv_planetable_clear):
	  Add some NULL pointer checks (possibly related to #385623).

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2006-12-12  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/apetag/gsttagdemux.c: (gst_tag_demux_chain_parse_tag),
	(gst_tag_demux_chain):
	* gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
	  In streaming mode, if the first buffer we get doesn't have an
	  offset, fix it up to be 0, otherwise trimming won't work later on
	  and we'll be typefinding application/x-id3, which may result in
	  decodebin plugging an endless number of id3demux elements as a
	  consequence. Fixes #385031.
	  
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2006-12-08  Jan Schmidt  <thaytan@mad.scientist.com>

	* sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_prepare):
	  Ignore the buffer_time the sound device reports. Turns out it is 
	  sometimes completely bogus and we're better off without it.

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2006-12-11  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
	(gst_matroska_demux_video_caps):
	* gst/matroska/matroska-ids.c:
	(gst_matroska_track_init_video_context):
	* gst/matroska/matroska-ids.h:
	  Try harder to extract the framerate for video tracks correctly and
	  save it directly instead of converting it back and forth a few
	  times. Mostly makes a difference for very small framerates (<1).
	  Fixes #380199.

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2006-12-11  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_init),
	(gst_gconf_audio_src_dispose), (do_toggle_element):
	* ext/gconf/gstgconfaudiosrc.h:
	  Remove gconf notify hook when the gconfaudiosrc element is
	  destroyed, otherwise the callback may be called on an
	  already-destroyed instance and bad things happen. Should fix
	  #378184.
	  Also ignore gconf key changes when the source is already running.

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2006-12-09  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Sebastian Dröge  <mail at slomosnail de>

	* gst/apetag/gstapedemux.c: (ape_demux_parse_tags):
	  We need to be able to read and parse any possible floating point string
	  format ("1,234" or "1.234") irrespective of the current locale. g_strod()
	  will parse the former only in certain locales though, so we really need
	  to canonicalise the separator to '.' and then use g_ascii_strtod() to
	  make sure we can parse either version at all times.
	  Fixes #382982 for real.

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2006-12-08  Jan Schmidt  <thaytan@mad.scientist.com>

	* sys/sunaudio/gstsunaudiomixerctrl.c:
	* sys/sunaudio/gstsunaudiosrc.c:

        Use the sunaudio debug category.

	* sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_finalize),
	(gst_sunaudiosink_class_init), (gst_sunaudiosink_init),
	(gst_sunaudiosink_set_property), (gst_sunaudiosink_get_property),
	(gst_sunaudiosink_open), (gst_sunaudiosink_close),
	(gst_sunaudiosink_prepare), (gst_sunaudio_sink_do_delay),
	(gst_sunaudiosink_write), (gst_sunaudiosink_delay),
	(gst_sunaudiosink_reset):
	* sys/sunaudio/gstsunaudiosink.h:

	Uses the sunaudio debug category for all debug output
 	Implements the _delay() callback to synchronise video playback better
 	Change the segtotal and segsize values back to the parent class 
          defaults (taken from buffer_time and latency_times of 200ms and 10ms 
          respectively)
	Measure the samples written to the device vs. played.
	Keep track of segments in the device by writing empty eof frames, and
	sleep using a GCond when we get too far ahead and risk overrunning the
	sink's ringbuffer.

	Fixes: #360673

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	Patch by: Sebastian Dröge  <mail at slomosnail de >

	* gst/audiofx/audiopanorama.c: (gst_audio_panorama_init),
	(gst_audio_panorama_set_caps), (gst_audio_panorama_transform):
	* gst/audiofx/audiopanorama.h:
	Fix audiopanorame with float samples. Fixes #383726.

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	* sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_reset):
	* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_open),
	(gst_sunaudiosrc_reset):

	Implement reset functions to unblock the src/sink more quickly on 
	state change requests.
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	* sys/sunaudio/gstsunaudiomixer.c:
	(gst_sunaudiomixer_change_state):
	Construct the correct mixer device name when the AUDIODEV env var
	is set.

	Patch by: Jerry Tan <jerry.tan at sun dot com>
	Fixes: #383596

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2006-12-08  Jan Schmidt  <thaytan@mad.scientist.com>

	* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_open):
	Apply patch to open the mixer control and set the MULTIPLE_OPEN
	ioctl. On solaris, the mixer device doesn't need opening non-blocking 
	- it can be opened by multiple processes by default, but needs the ioctl 	for multiple opens within 1 process.
	Patch by: Jerry Tan <jerry.tan at sun dot com>
	Fixes: #349015

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	* gst/smpte/gstmask.h:
	* gst/smpte/gstsmpte.c: (gst_smpte_class_init),
	(gst_smpte_setcaps), (gst_smpte_init), (gst_smpte_reset),
	(gst_smpte_collected), (gst_smpte_set_property),
	(gst_smpte_get_property), (gst_smpte_change_state), (plugin_init):
	* gst/smpte/gstsmpte.h:
	Port to 0.10 some more. 
	Added duration property to specify the duration of the transition.
	Make framerate a fraction.
	Deprecate fps property, we only use negotiated fps.
	Added docs.
	Fix collectpad usage.
	Reset state in READY.
	Send NEWSEGMENT event.
	Fix racy updates of object properties.
	Added debug category.
	Fixes #383323.

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	Patch by: Sjoerd Simons <sjoerd at luon dot net>

	* gst/videomixer/videomixer.c:
	(gst_videomixer_set_master_geometry),
	(gst_videomixer_pad_sink_setcaps), (gst_videomixer_collect_free):
	Don't reset xpos and ypos in the setcaps function because causes
	unexpected behaviour.
	Fixes #382179.

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	* gst/multipart/multipartmux.c: (gst_multipart_mux_compare_pads),
	(gst_multipart_mux_queue_pads), (gst_multipart_mux_collected):
	Keep track of the buffer timestamp in the collectdata member instead
	of modifying the buffer without making the metadata writable first.
	Fixes #382277.

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	Patch by: Rob Taylor <robtaylor at floopily dot org>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_start):
	If using multicast in udpsrc, bind to the multicast address rather than
	IN_ADDR_ANY.
	This allows the simultanous use of multiple udpsrcs listening on
	different multicat addresses. Without this all udpsrcs will receive all
	packets from all subscribed multicast addresses.
	Fixes #383001.

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	* ext/taglib/gstid3v2mux.cc:
	Don't attempt to write a NULL frame into the ID3 tag set when the 
	createFrame method returned NULL.
	Fixes: #381857
	Patch by: Jonathan Matthew <jonathan at 0kaolin wh9 net >

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	* gst/apetag/gstapedemux.c: (ape_demux_parse_tags):
	Use g_strtod() instead of sscanf to parse doubles, so that it will
	try parsing in the C locale if the current locale fails.
	Fixes: #382982
	Patch by: Sebastian Dröge  <mail at slomosnail de >

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2006-12-01  Jan Schmidt  <thaytan@mad.scientist.com>

	* win32/MANIFEST:
	Fix compilation on win32 under VS8
	Patch by: Sergey Scobich <sergey dot scobich at gmail dot com>
	Partially fixes #381175

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	* gst/avi/gstavimux.c:
	  accept all mpegversions,fixes #380825
	  spotted by: Jerome Alet  

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	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
	(gst_v4l2src_queue_frame), (gst_v4l2src_grab_frame),
	(gst_v4l2src_get_capture), (gst_v4l2src_set_capture),
	(gst_v4l2src_capture_init), (gst_v4l2src_buffer_finalize):
	  cleanup the error message a bit more

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2006-11-28  Wim Taymans  <wim@fluendo.com>

	* ext/libcaca/gstcacasink.c: (gst_cacasink_class_init):
	Fix width and height properties.

	* ext/libcaca/gstcacasink.h:
	Fix compilation on newer libcaca that require us to include a new
	header. Fixes #379918.

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2006-11-28  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open):
	* gst/rtsp/gstrtspsrc.h:
	* gst/rtsp/rtspext.h:
	* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_configure_stream),
	(rtsp_ext_wms_get_context):
	Add method so that extensions can choose to disable the setup of
	a stream.
	Make the WMS extension skip setup of x-wms-rtx streams. Fixes #377792.

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2006-11-27  Wim Taymans  <wim@fluendo.com>

	Patch by: Jonas Holmberg <jonas dot holmberg at axis dot com>

	* gst/multipart/multipartmux.c: (gst_multipart_mux_collected):
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	Push header in a separate buffer instead of memcpy:ing all data.
	Change LF => CRLF in headers.
	Move trailing LF to header. Fixes #379792.
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2006-11-27  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_chain):
	Small buffer overflow fix and improve debugging.

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2006-11-24  Stefan Kost  <ensonic@users.sf.net>

	* ext/esd/esdmon.h:
	* ext/esd/esdsink.h:
	  remove obsolete _factory_init protos

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2006-11-24  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_index_entry_for_time),
	(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
	(gst_avi_demux_peek_chunk), (gst_avi_demux_parse_subindex),
	(gst_avi_demux_read_subindexes_push),
	(gst_avi_demux_read_subindexes_pull), (gst_avi_demux_parse_stream),
	(gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
	(gst_avi_demux_sync), (gst_avi_demux_next_data_buffer),
	(gst_avi_demux_massage_index),
	(gst_avi_demux_calculate_durations_from_index),
	(gst_avi_demux_stream_header_pull), (gst_avi_demux_do_seek),
	(gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry),
	(gst_avi_demux_stream_data), (gst_avi_demux_loop):
	  remove dead code, tweak debugs statements, add comments, use
	  _uint64_scale instead _uint64_scale_int when using guint64 values,
	  small optimizations, reflow some error handling

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2006-11-22  Edward Hervey  <edward@fluendo.com>

	* po/.cvsignore:
	We never put .pot files in cvs. Let's ignore them all.

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2006-11-19  Tim-Philipp Müller  <tim at centricular dot net>

	* po/POTFILES.in:
	  ... but better exclude files that aren't disted.

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2006-11-19  Tim-Philipp Müller  <tim at centricular dot net>

	* po/POTFILES.in:
	  Add v4l2 source files to list of files with translations, so the
	  strings are actually extracted (however bad they still may be).

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2006-11-19  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/videobox/gstvideobox.c: (gst_video_box_class_init):
	  Minor clean-ups: const-ify static array, remove trailing comma from
	  last enum (gcc-2.9x trips over that), use GST_DEBUG_FUNCPTR.

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2006-11-19  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame):
	Make sure that g_free always gets called on the same pointer that was 
	returned by g_malloc.  Fixes #376594.
	Do not leak memory if decompressed size is wrong.
	Remove unneeded check of return value of g_malloc.
	Patch by: René Stadler <mail@renestadler.de>

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2006-11-18  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_capture_deinit):
	  Add missing curly brackets.

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2006-11-17  Edgard Lima <edgard.lima@indt.org.br>

	* sys/v4l2/v4l2src_calls.c:
	Fix capture_deinit.

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2006-11-16  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/matroska/matroska-mux.c: (gst_matroska_mux_class_init),
	(gst_matroska_mux_request_new_pad):
	  Use GST_DEBUG_FUNCPTR; activate request pad before returning it.

	* tests/check/elements/matroskamux.c: (setup_src_pad),
	(setup_sink_pad), (GST_START_TEST):
	Activate pads before using them.

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2006-11-16  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_stream_scan):
	  Initialise variable to get rid of bogus compiler warning.

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2006-11-16  Stefan Kost  <ensonic@users.sf.net>

	Patch by: Ville Syrjala <ville.syrjala@movial.fi>

	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph263ppay.c:
	  Specify H.263 variant and version in the caps (fixes #361637)

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2006-11-15  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/rtspconnection.c: (read_body):
	Don't set a data pointer to NULL and a size > 0 when we deal
	with empty packets.

	* gst/rtsp/rtspmessage.c: (rtsp_message_new_response),
	(rtsp_message_init_response), (rtsp_message_init_data),
	(rtsp_message_unset), (rtsp_message_free),
	(rtsp_message_take_body):
	Check that we can't create invalid empty packets. 

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2006-11-15  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Mark Nauwelaerts  <manauw at skynet be>

	* gst/matroska/matroska-mux.c: (gst_matroska_mux_add_interfaces),
	(gst_matroska_mux_class_init), (gst_matroska_pad_free),
	(gst_matroska_mux_reset), (gst_matroska_mux_handle_sink_event),
	(gst_matroska_mux_request_new_pad), (gst_matroska_mux_release_pad),
	(gst_matroska_mux_track_header), (gst_matroska_mux_start),
	(gst_matroska_mux_write_simple_tag), (gst_matroska_mux_finish):
	* gst/matroska/matroska-mux.h:
	  Add basic tag writing support; implement releasing pads (#374658).

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2006-11-15  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
	(gst_matroska_demux_audio_caps):
	  Handle opaque/unspecified A_AAC audio codec ID (fixes #374737).

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2006-11-14  David Schleef  <ds@schleef.org>

	* gst/matroska/matroska-mux.c: Add Dirac fourcc.

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2006-11-14  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Sergey Scobich  <sergey.scobich at gmail com>

	* win32/vs8/gst-plugins-good.sln:
	* win32/vs8/libgst1394.vcproj:
	* win32/vs8/libgstaasink.vcproj:
	* win32/vs8/libgstalaw.vcproj:
	* win32/vs8/libgstalpha.vcproj:
	* win32/vs8/libgstalphacolor.vcproj:
	* win32/vs8/libgstannodex.vcproj:
	* win32/vs8/libgstapetag.vcproj:
	* win32/vs8/libgstaudiofx.vcproj:
	* win32/vs8/libgstauparse.vcproj:
	* win32/vs8/libgstautodetect.vcproj:
	* win32/vs8/libgstavi.vcproj:
	* win32/vs8/libgstcacasink.vcproj:
	* win32/vs8/libgstcdio.vcproj:
	* win32/vs8/libgstcutter.vcproj:
	* win32/vs8/libgstdv.vcproj:
	* win32/vs8/libgsteffectv.vcproj:
	* win32/vs8/libgstflac.vcproj:
	* win32/vs8/libgstflxdec.vcproj:
	* win32/vs8/libgstgoom.vcproj:
	* win32/vs8/libgsticydemux.vcproj:
	* win32/vs8/libgstid3demux.vcproj:
	* win32/vs8/libgstjpeg.vcproj:
	* win32/vs8/libgstladspa.vcproj:
	* win32/vs8/libgstlevel.vcproj:
	* win32/vs8/libgstmatroska.vcproj:
	* win32/vs8/libgstmikmod.vcproj:
	* win32/vs8/libgstmng.vcproj:
	* win32/vs8/libgstmonoscope.vcproj:
	* win32/vs8/libgstmulaw.vcproj:
	* win32/vs8/libgstmultipart.vcproj:
	* win32/vs8/libgstpng.vcproj:
	* win32/vs8/libgstrtp.vcproj:
	* win32/vs8/libgstrtsp.vcproj:
	* win32/vs8/libgstshout2.vcproj:
	* win32/vs8/libgstsmpte.vcproj:
	* win32/vs8/libgstspeex.vcproj:
	* win32/vs8/libgsttaglib.vcproj:
	* win32/vs8/libgstudp.vcproj:
	* win32/vs8/libgstvideobalance.vcproj:
	* win32/vs8/libgstvideobox.vcproj:
	* win32/vs8/libgstvideoflip.vcproj:
	* win32/vs8/libgstvideomixer.vcproj:
	* win32/vs8/libgstwavenc.vcproj:
	* win32/vs8/libgstwavparse.vcproj:
	  Make end-of-line returns unixy, so that when the files are checked
	  out on win32 the line returns will be 0d 0a and not 0d 0d 0a.
	  Hopefully fixes #366492.

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	* gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
	Disable init_frames delay timestamp adjustment, it does not
	seem to be needed at all. Fixes #369621.

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	Patch by: Mark Nauwelaerts  <manauw at skynet be>

	* gst/videomixer/videomixer.c:
	(gst_videomixer_set_master_geometry),
	(gst_videomixer_pad_sink_setcaps), (gst_videomixer_class_init),
	(gst_videomixer_collect_free), (gst_videomixer_reset),
	(gst_videomixer_init), (gst_videomixer_finalize),
	(gst_videomixer_request_new_pad), (gst_videomixer_release_pad),
	(gst_videomixer_collected), (gst_videomixer_change_state):
	Fix memleak by unref'ing collectpads instance (when finalizing)
	Implement releasing a request pad. Fixes #374479.

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2006-11-10  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Sergey Scobich  <sergey.scobich at gmail com>

	* win32/vs8/gst-plugins-good.sln:
	* win32/vs8/libgst1394.vcproj:
	* win32/vs8/libgstaasink.vcproj:
	* win32/vs8/libgstalaw.vcproj:
	* win32/vs8/libgstalpha.vcproj:
	* win32/vs8/libgstalphacolor.vcproj:
	* win32/vs8/libgstannodex.vcproj:
	* win32/vs8/libgstapetag.vcproj:
	* win32/vs8/libgstaudiofx.vcproj:
	* win32/vs8/libgstauparse.vcproj:
	* win32/vs8/libgstautodetect.vcproj:
	* win32/vs8/libgstavi.vcproj:
	* win32/vs8/libgstcacasink.vcproj:
	* win32/vs8/libgstcdio.vcproj:
	* win32/vs8/libgstcutter.vcproj:
	* win32/vs8/libgstdv.vcproj:
	* win32/vs8/libgsteffectv.vcproj:
	* win32/vs8/libgstflac.vcproj:
	* win32/vs8/libgstflxdec.vcproj:
	* win32/vs8/libgstgoom.vcproj:
	* win32/vs8/libgsticydemux.vcproj:
	* win32/vs8/libgstid3demux.vcproj:
	* win32/vs8/libgstjpeg.vcproj:
	* win32/vs8/libgstladspa.vcproj:
	* win32/vs8/libgstlevel.vcproj:
	* win32/vs8/libgstmatroska.vcproj:
	* win32/vs8/libgstmikmod.vcproj:
	* win32/vs8/libgstmng.vcproj:
	* win32/vs8/libgstmonoscope.vcproj:
	* win32/vs8/libgstmulaw.vcproj:
	* win32/vs8/libgstmultipart.vcproj:
	* win32/vs8/libgstpng.vcproj:
	* win32/vs8/libgstrtp.vcproj:
	* win32/vs8/libgstrtsp.vcproj:
	* win32/vs8/libgstshout2.vcproj:
	* win32/vs8/libgstsmpte.vcproj:
	* win32/vs8/libgstspeex.vcproj:
	* win32/vs8/libgsttaglib.vcproj:
	* win32/vs8/libgstudp.vcproj:
	* win32/vs8/libgstvideobalance.vcproj:
	* win32/vs8/libgstvideobox.vcproj:
	* win32/vs8/libgstvideoflip.vcproj:
	* win32/vs8/libgstvideomixer.vcproj:
	* win32/vs8/libgstwavenc.vcproj:
	* win32/vs8/libgstwavparse.vcproj:
	  Add VS8 project files (note that many of the plugins in ext are
	  disabled by default). Fixes #366492.

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	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_grab_frame):
	  we do not translate debug messages

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	* gst/flx/gstflxdec.c: (gst_flxdec_class_init):
	  fix categorisation, make short desc more explicit, remove unused code
	  Fixes #372021

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	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtp/gstrtpamrpay.c:
	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtph263pdepay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtpmp2tdepay.c:
	* gst/rtp/gstrtpmp4gdepay.c:
	* gst/rtp/gstrtpmp4gpay.c:
	* gst/rtp/gstrtpmp4vdepay.c:
	* gst/rtp/gstrtpmp4vpay.c:
	* gst/rtp/gstrtpmpadepay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtppcmadepay.c:
	* gst/rtp/gstrtppcmapay.c:
	* gst/rtp/gstrtppcmudepay.c:
	* gst/rtp/gstrtppcmupay.c:
	* gst/rtp/gstrtpspeexdepay.c:
	* gst/rtp/gstrtpspeexpay.c:
	* gst/rtp/gstrtpsv3vdepay.c:
	Fix element descriptions.

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	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_handle_buffer):
	Fix description.
	Small cleanup in the payloader.

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	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_base_init),
	(gst_rtp_theora_depay_class_init), (gst_rtp_theora_depay_init),
	(gst_rtp_theora_depay_finalize),
	(gst_rtp_theora_depay_parse_configuration),
	(gst_rtp_theora_depay_setcaps),
	(gst_rtp_theora_depay_switch_codebook),
	(gst_rtp_theora_depay_process),
	(gst_rtp_theora_depay_set_property),
	(gst_rtp_theora_depay_get_property),
	(gst_rtp_theora_depay_change_state),
	(gst_rtp_theora_depay_plugin_init):
	* gst/rtp/gstrtptheoradepay.h:
	* gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_base_init),
	(gst_rtp_theora_pay_class_init), (gst_rtp_theora_pay_init),
	(gst_rtp_theora_pay_setcaps), (gst_rtp_theora_pay_reset_packet),
	(gst_rtp_theora_pay_init_packet),
	(gst_rtp_theora_pay_flush_packet),
	(gst_rtp_theora_pay_finish_headers), (gst_rtp_theora_pay_parse_id),
	(gst_rtp_theora_pay_handle_buffer),
	(gst_rtp_theora_pay_plugin_init):
	* gst/rtp/gstrtptheorapay.h:
	Add theora pay/depayloaders.

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	* gst/rtp/Makefile.am:
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	We depend on gsttag to generate the vorbis comments.
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	* gst/rtp/gstrtpvorbisdepay.c:
	(gst_rtp_vorbis_depay_parse_configuration),
	(gst_rtp_vorbis_depay_setcaps),
	(gst_rtp_vorbis_depay_switch_codebook),
	(gst_rtp_vorbis_depay_process):
	* gst/rtp/gstrtpvorbisdepay.h:
	Parse configuration string in the depayloader.
	Implement selecting and switching to a new codebook.
	Receiving vorbis over RTP now works.

	* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_reset_packet),
	(gst_rtp_vorbis_pay_init_packet),
	(gst_rtp_vorbis_pay_finish_headers),
	(gst_rtp_vorbis_pay_handle_buffer):
	* gst/rtp/gstrtpvorbispay.h:
	Set timestamps on outgoing buffers and RTP packets.
	Fix configuration string, prepend number of Packet headers.
	Fix encoding of ident string.
	Add delivery-method to caps.
	Streaming vorbis over RTP now works.

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	* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_setcaps),
	(gst_rtp_vorbis_pay_finish_headers), (gst_rtp_vorbis_pay_parse_id),
	(gst_rtp_vorbis_pay_handle_buffer):
	* gst/rtp/gstrtpvorbispay.h:
        Generate a valid configuration string in the caps based on the
        vorbis headers.

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	* ext/cdio/gstcdio.c: (gst_cdio_get_cdtext):
	* ext/cdio/gstcdio.h:
	* ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_open):
	  Move CD-TEXT utility function into common file so it can also be
	  used by a future cdioparanoiasrc.

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	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2xoverlay.c:
	* sys/v4l2/v4l2_calls.c:
	* sys/v4l2/v4l2src_calls.c:
	Improved comments in ELEMENT_ERROR/WARNING and added "#if 0" to
	xoverlay code that is still not implemented.

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	* gst/id3demux/id3v2frames.c: (parse_picture_frame):
	  We require a -base more recent than 0.10.9, so it's safe to use
	  GST_TYPE_TAG_IMAGE_TYPE unconditionally now.

	* ext/dv/gstdvdec.c: (gst_dvdec_sink_event):
	* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_sink_event):
	  Use _newsegment_full() now that we depend on a recent enough core.

	* gst/wavparse/gstwavparse.c:
	  Remove cruft that we don't need any longer now that we depend on
	  a recent enough -base.

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	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_init),
	(gst_rtpilbcpay_setcaps):
	Fix and activate ILBC pay and depayloaders. Fixes #368162.

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	* ext/speex/gstspeexdec.c: (speex_dec_convert),
	(speex_dec_sink_event), (speex_dec_chain_parse_header):
	Some small cleanups, use _scale.

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	* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query):
	Use higher precision scale function.

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	Patch by: Michal Benes  <michal dot benes at itonis tv>

	* gst/matroska/matroska-demux.c: (gst_matroska_demux_encoding_cmp),
	(gst_matroska_demux_read_track_encodings),
	(gst_matroska_decode_buffer):
	  Fix several issues with encoded/compressed/encrypted/signed tracks;
	  also, remove superfluous newline characters from some debug
	  statements. (#366155)

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2006-10-30  Wim Taymans  <wim@fluendo.com>

	* ext/jpeg/gstjpegenc.c: (gst_jpegenc_getcaps):
	* ext/jpeg/gstsmokedec.c: (gst_smokedec_class_init),
	(gst_smokedec_init), (gst_smokedec_finalize), (gst_smokedec_chain),
	(gst_smokedec_change_state):
	* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_class_init),
	(gst_smokeenc_init), (gst_smokeenc_finalize),
	(gst_smokeenc_getcaps), (gst_smokeenc_setcaps),
	(gst_smokeenc_resync), (gst_smokeenc_chain),
	(gst_smokeenc_set_property), (gst_smokeenc_get_property),
	(gst_smokeenc_change_state):
	Various cleanups, capsnego and leak fixes.

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2006-10-30  Wim Taymans  <wim@fluendo.com>

	Patch by: Mark Nauwelaerts  <manauw at skynet be>

	* gst/videomixer/videomixer.c: (gst_videomixer_update_queues):
	Fix videomixer so that it can handle any combination of framerates.
	Fixes #367221.

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2006-10-28  Wim Taymans  <wim@fluendo.com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
	(gst_avi_demux_parse_file_header),
	(gst_avi_demux_stream_init_push), (gst_avi_demux_parse_stream),
	(gst_avi_demux_stream_header_push), (gst_avi_demux_stream_data),
	(gst_avi_demux_chain):
	Fix position query for audio. also fixes timestamps in streaming
	mode and bug #364958.
	Small cleanups.

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2006-10-27  Wim Taymans  <wim@fluendo.com>

	* ext/libpng/gstpngenc.c: (gst_pngenc_setcaps), (gst_pngenc_chain):
	* ext/libpng/gstpngenc.h:
	Fix strides. Fixes #364856.
	Cleanup capsnego.
	Set caps on outgoing buffers.

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2006-10-18  Wim Taymans  <wim@fluendo.com>

	Patch by: Ville Syrjala <ville dot syrjala at movial dot fi>

	* gst/rtp/gstrtpgsmpay.c:
	* gst/rtp/gstrtph263pay.c:
	* gst/rtp/gstrtpmpapay.c:
	* gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_flush),
	(gst_rtp_pcma_pay_handle_buffer):
	* gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_flush):
	Add static payload numbers in addition to the dynamic ones.
	Fixes #361639.

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2006-10-18  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtsp_lower_trans_get_type),
	(gst_rtspsrc_class_init), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_open),
	(gst_rtspsrc_uri_get_protocols), (gst_rtspsrc_uri_set_uri):
	* gst/rtsp/rtspconnection.c: (rtsp_connection_create):
	* gst/rtsp/rtspdefs.h:
	* gst/rtsp/rtspurl.c: (rtsp_url_parse):
	* gst/rtsp/rtspurl.h:
	Reuse already existing enum for lower transport.
	Add rtspt and rtspu protocols.
	Send redirect to rtspt when udp times out.

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	* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
	(gst_wavparse_stream_data):
	Fix seeking some more, mostly for speed changes.

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	* sys/v4l2/gstv4l2tuner.c:
	* sys/v4l2/gstv4l2tuner.h:
	  Fix _set_channel(): remove useless g_object_notify() for "channel"
	  property that doesn't exist any longer and therefore now also
	  useless redirect (#338818).

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	* sys/oss/gstosssink.c: (gst_oss_sink_prepare):
	Some drivers do not support unsetting the non-blocking flag once the
	device is opened. In those cases, close/open the device in
	non-blocking mode. Fixes #362673.

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	* sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_set_fps),
	(gst_v4l2src_get_fps):
	  dear stefan, framespersecond is not frameperiod, reverting but adding
	  comment

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	* sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_set_fps),
	(gst_v4l2src_get_fps):
	  Numerator is numerator and denominator is denominator. Say that aloud
	  5 times and retry after next beer.

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2006-10-16  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Josep Torra Valles  <josep at fluendo com>

	* ext/cairo/gsttimeoverlay.c: (gst_cairo_time_overlay_transform):
	* ext/esd/esdsink.c: (gst_esdsink_write):
	* ext/flac/gstflacdec.c: (gst_flac_dec_length),
	(gst_flac_dec_read_seekable), (gst_flac_dec_chain),
	(gst_flac_dec_send_newsegment):
	* ext/flac/gstflacenc.c: (gst_flac_enc_seek_callback),
	(gst_flac_enc_tell_callback):
	* ext/jpeg/smokecodec.c: (find_best_size), (smokecodec_encode),
	(smokecodec_parse_header), (smokecodec_decode):
	* gst/avi/gstavimux.c: (gst_avi_mux_write_avix_index):
	* gst/debug/efence.c: (gst_fenced_buffer_alloc):
	* gst/goom/Makefile.am:
	* gst/goom/gstgoom.c:
	* gst/icydemux/gsticydemux.c: (gst_icydemux_typefind_or_forward):
	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/rtspconnection.c: (rtsp_connection_read):
	* gst/udp/gstudpsink.c:
	* gst/udp/gstudpsrc.c:
	* gst/wavparse/gstwavparse.c: (gst_wavparse_change_state):
	* sys/sunaudio/gstsunaudiomixertrack.h:
	  Fix a bunch of problems discovered by the Forte compiler, mostly type
	  mixups and pointer arithmetics with void pointers. Fixes #362603.

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2006-10-12  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/speex/gstspeex.c: (plugin_init):
	* ext/speex/gstspeexenc.c: (gst_speex_enc_get_formats),
	(gst_speex_enc_setup_interfaces), (gst_speex_enc_base_init),
	(gst_speex_enc_class_init), (gst_speex_enc_finalize),
	(gst_speex_enc_sink_setcaps), (gst_speex_enc_convert_src),
	(gst_speex_enc_convert_sink), (gst_speex_enc_get_query_types),
	(gst_speex_enc_src_query), (gst_speex_enc_sink_query),
	(gst_speex_enc_init), (gst_speex_enc_create_metadata_buffer),
	(gst_speex_enc_set_last_msg), (gst_speex_enc_setup),
	(gst_speex_enc_buffer_from_data), (gst_speex_enc_push_buffer),
	(gst_speex_enc_set_header_on_caps), (gst_speex_enc_sinkevent),
	(gst_speex_enc_chain), (gst_speex_enc_get_property),
	(gst_speex_enc_set_property), (gst_speex_enc_change_state):
	* ext/speex/gstspeexenc.h:
	  Miscellaneous clean-ups, among other things: speexenc => enc to
	  enhance code readability; change speexenc => speex_enc; in chain
	  function unref input buffer in case of error; take reference in
	  event function; use boilerplate macro; use gst_pad_query_peer_*
	  convenience functions.

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2006-10-12  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/speex/gstspeexenc.c: (gst_speexenc_finalize),
	(gst_speexenc_set_last_msg), (gst_speexenc_setup),
	(gst_speexenc_set_header_on_caps):
	  Fix some mem leaks.

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2006-10-11  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/URLS:
	Added some other URL.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp),
	(gst_rtspsrc_handle_request), (gst_rtspsrc_send),
	(gst_rtspsrc_open), (gst_rtspsrc_play),
	(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	* gst/rtsp/gstrtspsrc.h:
	Work on fallback to TCP connection when the UDP socket times out.
	Handler server requests, just reply with OK for now.

	* gst/rtsp/rtspdefs.c: (rtsp_strresult):
	* gst/rtsp/rtspdefs.h:
	Added some more Real extension headers.

	* gst/rtsp/rtspurl.c: (rtsp_url_parse):
	Fix parsing of urls with a ':' that is not part of the hostname:port
	part of the url.

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2006-10-11  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/apetag/gsttagdemux.c: (gst_tag_demux_add_srcpad):
	* gst/icydemux/gsticydemux.c: (gst_icydemux_add_srcpad):
	* gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad):
	  Activate pad before adding it to the already-running element.

	* tests/check/elements/icydemux.c: (icydemux_found_pad):
	  Activate newly-created pad too.

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2006-10-11  Wim Taymans  <wim@fluendo.com>

	Patch by: Sebastien Cote <sebas642 at yahoo dot ca>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
	(gst_udpsrc_finalize), (gst_udpsrc_create), (gst_udpsrc_set_uri),
	(gst_udpsrc_start):
	Fix some leaks in caps and uris. Fixes #361252.

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2006-10-10  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/wavparse/Makefile.am:
	  Fix copy'n'paste-o (spotted by Mark Nauwelaerts, #341489).

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2006-10-09  Jan Schmidt  <thaytan@mad.scientist.com>

	* sys/v4l2/gstv4l2xoverlay.c:
	* sys/v4l2/gstv4l2xoverlay.h:
	Fix build as per the patch in #338818 comment 36.

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2006-10-07  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport):
	  Activate pads before adding them to the source.

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2006-10-06  Wim Taymans  <wim@fluendo.com>

	* ext/dv/gstdvdemux.c: (gst_dvdemux_add_pads), (gst_dvdemux_chain):
	* gst/auparse/gstauparse.c: (gst_au_parse_add_srcpad):
	Activate pads before adding.

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2006-10-06  Wim Taymans  <wim@fluendo.com>

	* gst/multipart/multipartdemux.c: (gst_multipart_demux_init),
	(gst_multipart_find_pad_by_mime):
	Activate pads before adding.

	* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init):
	BOILERPLATE sets parent_class for us.

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2006-10-06  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
	(gst_rtspsrc_class_init), (gst_rtspsrc_init),
	(gst_rtspsrc_create_stream), (gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_alloc_udp_ports),
	(gst_rtspsrc_stream_configure_transport), (find_stream_by_channel),
	(gst_rtspsrc_push_event), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_create_transports_string),
	(gst_rtspsrc_configure_transports), (gst_rtspsrc_open),
	(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	* gst/rtsp/gstrtspsrc.h:
	Rework how the transport string is constructed, try to share channels
	and udp ports.
	Make most of the stuff less dependant on RTP as we are also going to use
	it for RDT.
	Add support for transport specific session managers.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_flush):
	Implement _flush().

	* gst/rtsp/rtspdefs.c: (rtsp_strresult):
	* gst/rtsp/rtspdefs.h:
	Add generic error return code.

	* gst/rtsp/rtspext.h:
	Add support for pluggable tranport strings.

	* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_before_send),
	(rtsp_ext_wms_after_send), (rtsp_ext_wms_parse_sdp),
	(rtsp_ext_wms_get_context):
	Detect WMServer and activate the extension.

	* gst/rtsp/rtsptransport.c: (rtsp_transport_get_mime),
	(rtsp_transport_get_manager), (rtsp_transport_parse):
	* gst/rtsp/rtsptransport.h:
	Added methods to get mime/manager for certain transports.

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2006-10-05  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/cairo/gsttimeoverlay.c:
	(gst_cairo_time_overlay_update_font_height):
	* ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_transform_caps):
	* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_parse_image_data):
	* ext/jpeg/gstjpegenc.c: (gst_jpegenc_chain):
	* ext/jpeg/gstsmokedec.c: (gst_smokedec_chain):
	* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_chain):
	* ext/libpng/gstpngdec.c: (user_endrow_callback):
	* gst/auparse/gstauparse.c: (gst_au_parse_parse_header):
	* gst/avi/gstavidemux.c: (gst_avi_demux_parse_superindex),
	(gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream),
	(gst_avi_demux_stream_data):
	* gst/cutter/gstcutter.c: (gst_cutter_chain):
	* gst/debug/efence.c: (gst_efence_buffer_alloc),
	(gst_fenced_buffer_copy):
	* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame):
	* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream):
	* gst/matroska/matroska-mux.c: (gst_matroska_mux_start):
	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send),
	(gst_rtspsrc_handle_message):
	* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
	* sys/ximage/ximageutil.c: (ximageutil_xcontext_get):
	  Printf format fixes.

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2006-10-04  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/Makefile.am:
	Dist new .h file too.

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2006-10-04  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/Makefile.am:
	* gst/rtsp/gstrtpdec.c: (gst_rtpdec_getcaps),
	(gst_rtpdec_chain_rtp), (gst_rtpdec_chain_rtcp):
	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	(gst_rtspsrc_finalize), (gst_rtspsrc_create_stream),
	(gst_rtspsrc_parse_rtpmap),
	(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_send),
	(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
	(gst_rtspsrc_play), (gst_rtspsrc_handle_message):
	* gst/rtsp/gstrtspsrc.h:
	* gst/rtsp/rtspdefs.c: (rtsp_strresult):
	* gst/rtsp/rtspdefs.h:
	* gst/rtsp/rtspext.h:
	* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp),
	(rtsp_ext_wms_get_context):
	* gst/rtsp/rtspextwms.h:
	* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
	(rtsp_transport_parse):
	* gst/rtsp/rtsptransport.h:
	Factor out extension in separate module.
	Fix getcaps to filter against the padtemplate.
	Use Content-Base if the server gives one.
	Rework the transport parsing a bit for future extensions.
	Added some Real Header field definitions.

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	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	  added v4l2 stubs
	* gst-plugins-good.spec.in:
	  add v4l2

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2006-10-04  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/apetag/gstapedemux.c: (ape_demux_parse_tags):
	  Extract disc/album/medium number and count and try harder
	  to extract track number/count.

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	* configure.ac:
	* sys/Makefile.am:
	  add build stuff for v4l2, needs --enable-experimental until
	  the last bits are resolved

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2006-09-29  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	  Disable autodetect test temporarily, so that the build bots
	  update -bad and the ranks of unreliable video sinks in there.

	* tests/check/elements/autodetect.c: (GST_START_TEST):
	  Skip test if no usable videosink is found.

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2006-09-29  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/URLS:
	Add some more URLs.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
	(gst_rtspsrc_init), (gst_rtspsrc_finalize),
	(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
	(gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
	(gst_rtspsrc_loop), (gst_rtspsrc_send),
	(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
	(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
	(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	* gst/rtsp/gstrtspsrc.h:
	Add timeout property to control UDP timeouts.
	Fix error messages.
	Also start a loop function when operating in UDP mode so that we can
	do some more stuff async.
	Handle element messages from udpsrc to detect timeouts. If a timeout
	happens we currently generate an error.
	API: rtspsrc::timeout property.

	* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
	(gst_udpsrc_create):
	Really implement the timeout in microseconds and not milliseconds.

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2006-09-29  Wim Taymans  <wim@fluendo.com>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
	(gst_udpsrc_create), (gst_udpsrc_set_property),
	(gst_udpsrc_get_property), (gst_udpsrc_unlock), (gst_udpsrc_stop):
	* gst/udp/gstudpsrc.h:
	Added property to post a message on timeout.
	Updated docs.
	When restarting the select, initialize the fdsets again.
	Init control sockets so we don't accidentally close a random socket.
	API: GstUDPSrc::timeout property

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2006-09-29  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type):
	Fix flag registration.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_read):
	Reading 0 also means 'no more commands'

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2006-09-29  Wim Taymans  <wim@fluendo.com>

	Patch by: Antoine Tremblay <hexa00 at gmail dot com>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
	Fix possible infinite loop when shutting down, a read can also return
	0 to indicate no more messages are available. Fixes #358156.

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2006-09-25  Wim Taymans  <wim@fluendo.com>

	* gst/autodetect/gstautoaudiosink.c:
	(gst_auto_audio_sink_base_init), (gst_auto_audio_sink_class_init),
	(gst_auto_audio_sink_find_best):
	* gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_detect):
	Small cleanups.
	don't try to set "sync" property when it is not available.

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2006-09-25  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/alpha/gstalpha.c:
	* gst/rtp/gstrtpamrdepay.c:
	* gst/rtsp/gstrtspsrc.c:
	* gst/udp/gstudpsrc.c:
	* gst/videomixer/videomixer.c:
	  Include stdlib.h in some more places, makes things compile
	  with uClibc and -Werror (#357592).

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2006-09-25  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/jpeg/gstjpegdec.c:
	  Set minimum height to 8 (from 16), our code should handle
	  that fine. Some of the buttons on the apple trailer site
	  are apparently only 15 pixels high (see #357470).

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2006-09-23  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop), (gst_rtspsrc_send),
	(gst_rtspsrc_open):
	* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	(rtsp_connection_connect), (rtsp_connection_read), (read_body),
	(rtsp_connection_receive):
	* gst/rtsp/rtspdefs.c: (rtsp_strresult):
	* gst/rtsp/rtspdefs.h:
	Improve error reporting.

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2006-09-23  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstasteriskh263.c: (gst_asteriskh263_plugin_init):
	* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_plugin_init):
	* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_plugin_init):
	* gst/rtp/gstrtpdepay.c:
	* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_plugin_init):
	* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_plugin_init):
	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_plugin_init):
	* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_plugin_init):
	* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps),
	(gst_rtp_mp2t_depay_plugin_init):
	* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_plugin_init):
	* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_plugin_init):
	* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_plugin_init):
	* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_plugin_init):
	* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_plugin_init):
	* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_plugin_init):
	* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_plugin_init):
	* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_plugin_init):
	Fix klass typos.
	Mark RANK_MARGINAL, decodebin can handle the depayloaders fine.

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2006-09-22  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  Need  -base CVS for gst_base_rtp_depayload_push_ts().

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2006-09-22  Wim Taymans  <wim@fluendo.com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index):
	Don't check for a tag that is never there and check if we read the
	correct tag. Fixes seeking again.
	We must post an error when all pads are unlinked.

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2006-09-22  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_process):
	* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_setcaps),
	(gst_rtp_vorbis_pay_reset_packet),
	(gst_rtp_vorbis_pay_init_packet),
	(gst_rtp_vorbis_pay_flush_packet), (gst_rtp_vorbis_pay_parse_id),
	(gst_rtp_vorbis_pay_handle_buffer):
	More fixage, set endoder-params correctly in the payloader.

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2006-09-22  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/autodetect/gstautoaudiosink.c:
	(gst_auto_audio_sink_base_init):
	* gst/autodetect/gstautovideosink.c:
	(gst_auto_video_sink_base_init):
	  Make static pad templates static to appease valgrind's leak
	  detector.

	* tests/check/Makefile.am:
	* tests/check/elements/.cvsignore:
	* tests/check/elements/autodetect.c: (GST_START_TEST),
	(autodetect_suite):
	  Add simple test for the ghostpad lockup on shutdown fixed in core
	  CVS (audio bit disabled because it would need dozens of alsa
	  suppressions and I'm too lazy to add those now).

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2006-09-22  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_change_state):
	* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_class_init):
	Small cleanups.

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_base_init),
	(gst_rtp_vorbis_depay_class_init), (gst_rtp_vorbis_depay_init),
	(gst_rtp_vorbis_depay_finalize), (gst_rtp_vorbis_depay_setcaps),
	(gst_rtp_vorbis_depay_process),
	(gst_rtp_vorbis_depay_set_property),
	(gst_rtp_vorbis_depay_get_property),
	(gst_rtp_vorbis_depay_change_state),
	(gst_rtp_vorbis_depay_plugin_init):
	* gst/rtp/gstrtpvorbisdepay.h:
	* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_base_init),
	(gst_rtp_vorbis_pay_class_init), (gst_rtp_vorbis_pay_init),
	(gst_rtp_vorbis_pay_setcaps), (gst_rtp_vorbis_pay_init_packet),
	(gst_rtp_vorbis_pay_flush_packet),
	(gst_rtp_vorbis_pay_append_buffer),
	(gst_rtp_vorbis_pay_handle_buffer),
	(gst_rtp_vorbis_pay_plugin_init):
	* gst/rtp/gstrtpvorbispay.h:
	Add experimental vorbis pay and depayloaders.

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2006-09-21  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_parse_audio_config):
	Fix profile-level-id parsing and setup.

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2006-09-21  Wim Taymans  <wim@fluendo.com>

	* gst/udp/README:
	* gst/udp/gstudpsrc.c: (gst_udpsrc_set_property):
	Update README, simple cleanup.

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2006-09-21  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/README:
	Update README with some examples.

	* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_init),
	(gst_rtp_mp4g_pay_finalize), (gst_rtp_mp4g_pay_parse_audio_config),
	(gst_rtp_mp4g_pay_parse_video_config), (gst_rtp_mp4g_pay_new_caps),
	(gst_rtp_mp4g_pay_setcaps):
	* gst/rtp/gstrtpmp4gpay.h:
	Make optional RTP parameters of type STRING, as required by the
	application/x-rtp caps specification.

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	* gst/rtp/gstrtph263pdepay.c:
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	* gst/rtp/gstrtph263ppay.c:
	Correctly calculate size of each H263+ RTP buffer taking into account MTU and
	RTP header.

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2006-09-20  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/Makefile.am:
	And makefile too.

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2006-09-20  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpasfdepay.c: (gst_rtp_asf_depay_base_init),
	(gst_rtp_asf_depay_class_init), (gst_rtp_asf_depay_init),
	(decode_base64), (gst_rtp_asf_depay_setcaps),
	(gst_rtp_asf_depay_process), (gst_rtp_asf_depay_set_property),
	(gst_rtp_asf_depay_get_property), (gst_rtp_asf_depay_change_state),
	(gst_rtp_asf_depay_plugin_init):
	* gst/rtp/gstrtpasfdepay.h:
	Added preliminary ASF depayloader.

	* gst/rtp/gstrtph264depay.c: (decode_base64):
	Fix base64 decoding.

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2006-09-20  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/URLS:
	Added some test URLS.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
	(gst_rtspsrc_loop), (gst_rtspsrc_open):
	* gst/rtsp/gstrtspsrc.h:
	When creating streams, give access to the complete SDP.
	Fix some leaks.
	Collect and merge global stream properties in stream caps.
	Preliminary support for WMServer.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	(rtsp_connection_connect), (rtsp_connection_read), (read_body),
	(rtsp_connection_receive):
	* gst/rtsp/rtspconnection.h:
	Make connection interruptable.
	Refactor to make it reconnectable.
	Don't fail on short reads when reading data packets.

	* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_set_port),
	(rtsp_url_get_port):
	* gst/rtsp/rtspurl.h:
	Add methods for getting/setting the port.

	* gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n),
	(sdp_message_get_attribute_val), (sdp_media_get_attribute),
	(sdp_media_get_attribute_val_n), (sdp_media_get_attribute_val),
	(sdp_media_get_format), (sdp_parse_line),
	(sdp_message_parse_buffer):
	Fix headers. 
	Add methods for getting multiple attributes with the same name.
	Increase buffer size when parsing.
	Fix parsing of a=foo fields.

	* gst/rtsp/test.c: (main):
	Update to new connection API.

	* gst/rtsp/rtspmessage.c: (rtsp_message_new_response),
	(rtsp_message_init_response), (rtsp_message_init_data),
	(rtsp_message_unset), (rtsp_message_free), (rtsp_message_dump):
	* gst/rtsp/rtspmessage.h:
	* gst/rtsp/rtsptransport.c: (rtsp_transport_free):
	* gst/rtsp/rtsptransport.h:
	* gst/rtsp/sdp.h:
	* gst/rtsp/sdpmessage.h:
	* gst/rtsp/gstrtsp.c:
	* gst/rtsp/gstrtsp.h:
	* gst/rtsp/gstrtpdec.c:
	* gst/rtsp/gstrtpdec.h:
	* gst/rtsp/rtsp.h:
	* gst/rtsp/rtspdefs.c:
	* gst/rtsp/rtspdefs.h:
	Dual licensed under MIT and LGPL now.

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2006-09-19  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (find_stream_by_pt),
	(gst_rtspsrc_create_stream), (gst_rtspsrc_free_stream),
	(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
	(gst_rtspsrc_stream_configure_transport), (find_stream_by_channel),
	(gst_rtspsrc_push_event), (gst_rtspsrc_loop), (gst_rtspsrc_send),
	(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
	(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
	* gst/rtsp/gstrtspsrc.h:
	Reorganize stream parsing and creation.
	Detect container formats in interleaved mode.
	Keep more state about the streams.
	Assume a server also supports PLAY if it does not say.
	Add unicast and interleaved properties to TCP transport requests to make
	some servers happy (WMServer).

	* gst/rtsp/sdpmessage.h:
	Add some defines for the standard Bandwidth types.

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2006-09-19  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/test.c: (main):
	Fix build.

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2006-09-19  Wim Taymans  <wim@fluendo.com>

	* gst/wavparse/gstwavparse.c:
	Add ms-gsm to the src template.

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2006-09-18  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_send), (gst_rtspsrc_parse_methods),
	(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play),
	(gst_rtspsrc_pause), (gst_rtspsrc_change_state),
	(gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri):
	* gst/rtsp/gstrtspsrc.h:
	Small cleanups, added documentation.
	Try to clean up the requests and responses.
	Refactor parsing the supported methods.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_open),
	(rtsp_connection_create), (rtsp_connection_send),
	(parse_response_status), (parse_request_line),
	(rtsp_connection_receive), (rtsp_connection_close),
	(rtsp_connection_free):
	* gst/rtsp/rtsptransport.c: (rtsp_transport_new),
	(rtsp_transport_init), (rtsp_transport_parse),
	(rtsp_transport_free):
	* gst/rtsp/rtspurl.c: (rtsp_url_parse):
	* gst/rtsp/sdpmessage.c: (sdp_message_new), (sdp_message_init),
	(sdp_message_clean), (sdp_message_free), (sdp_media_new),
	(sdp_media_init), (sdp_message_parse_buffer), (sdp_message_dump):
	Use g_return_val some more.

	* gst/rtsp/rtspdefs.h:
	Add more enum values to track initial states.

	* gst/rtsp/rtspmessage.c: (rtsp_message_new_request),
	(rtsp_message_init_request), (rtsp_message_new_response),
	(rtsp_message_init_response), (rtsp_message_init_data),
	(rtsp_message_unset), (rtsp_message_free),
	(rtsp_message_add_header), (rtsp_message_remove_header),
	(rtsp_message_get_header), (rtsp_message_set_body),
	(rtsp_message_take_body), (rtsp_message_get_body),
	(rtsp_message_steal_body), (rtsp_message_dump):
	* gst/rtsp/rtspmessage.h:
	Reorder arguments, object goes as the first one.
	Use g_return_val some more.

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2006-09-18  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_base_init),
	(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_push_event),
	(gst_rtspsrc_loop), (gst_rtspsrc_uri_set_uri):
	* gst/rtsp/gstrtspsrc.h:
	Export sometimes source pad with correct caps on the template, create
	the ghostpad from the template.
	Remove RTCP template as we never expose RTCP.
	Protect against invalid body size.
	Avoid memcpy when creating the output buffer.
	Properly post an error and send EOS when the loop function is shut down.

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2006-09-18  Wim Taymans  <wim@fluendo.com>

	Based on patch by: Lutz Mueller <lutz at topfrose dot de>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
	(gst_rtspsrc_init), (gst_rtspsrc_set_property), (gst_rtspsrc_open),
	(gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri):
	* gst/rtsp/gstrtspsrc.h:
	Make sure we can never set an invalid location.

	* gst/rtsp/rtspmessage.c: (rtsp_message_steal_body):
	* gst/rtsp/rtspmessage.h:
	Added _steal_body method for future use.

	* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_free):
	Make freeing of NULL url return immediatly.

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2006-09-18  Wim Taymans  <wim@fluendo.com>

	Based on patch by: Lutz Mueller <lutz at topfrose dot de>

	* gst/rtsp/gstrtspsrc.c: (_do_init), (gst_rtspsrc_class_init),
	(gst_rtspsrc_init), (gst_rtspsrc_stream_setup_rtp),
	(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_play),
	(gst_rtspsrc_change_state):
	* gst/rtsp/gstrtspsrc.h:
	Use boilerplate.
	Make rtspsrc subclass GstBin to make state changes easier.
	Add Range header field on the PLAY request.

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2006-09-18  Wim Taymans  <wim@fluendo.com>

	Based on patch by: Thijs Vermeir <thijs dot vermeir at barco dot com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state),
	(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
	(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open),
	(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause):
	* gst/rtsp/rtspconnection.c: (inet_aton):
	Small cleanups.
	when multicast is selected as the transport, create UDP sources and
	connect to the multicast group.
	Move parsing and setting of caps to a common place.
	Fixes #349894.

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2006-09-17  Stefan Kost  <ensonic@users.sf.net>

	* ext/flac/gstflactag.c:
	* gst/alpha/gstalpha.c:
	* gst/debug/breakmydata.c:
	* gst/debug/negotiation.c:
	* gst/debug/testplugin.c:
	* gst/effectv/gstaging.c:
	* gst/effectv/gstdice.c:
	* gst/effectv/gstedge.c:
	* gst/effectv/gstquark.c:
	* gst/effectv/gstrev.c:
	* gst/effectv/gstshagadelic.c:
	* gst/effectv/gstvertigo.c:
	* gst/effectv/gstwarp.c:
	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartmux.c:
	* gst/videobox/gstvideobox.c:
	* gst/videofilter/gstgamma.c:
	* gst/videofilter/gstvideotemplate.c:
	* gst/videomixer/videomixer.c:
	* sys/sunaudio/gstsunaudiosrc.h:
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	More G_OBJECT macro fixing.
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2006-09-16  Wim Taymans  <wim@fluendo.com>

	Patch by: Yves Lefebvre <ivanohe at abacom dot com>

	* gst/avi/gstavimux.c: (gst_avi_mux_stop_file):
	Correctly set the dwLength in strh.
	With this patch, the file duration is now displayed correctly in window
	media player and the AVI plays completely. Fixes #356147

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2006-09-15  Wim Taymans  <wim@fluendo.com>

	Patch by: Darren Kenny <darren dot kenny at sun dot com>

	* sys/sunaudio/gstsunaudiomixerctrl.c:
	(gst_sunaudiomixer_ctrl_build_list):
	Set the output track as the MASTER so that the gnome-settings-daemon
	keybindings for changing the volume using the keyboard works.
	Fixes #356142.

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2006-09-15  Wim Taymans  <wim@fluendo.com>

	* gst/multipart/multipartdemux.c: (gst_multipart_demux_chain):
	Fix documentation, it is not possible to control the framerate of jpegdec
	using filtered caps yet. Fixes #355210.
	Return the downstream GstFlowReturn instead of GST_FLOW_OK so that we
	stop when there is an error.

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2006-09-14  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/apetag/gsttagdemux.c: (gst_tag_demux_chain_parse_tag):
	* gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
	  Don't interpret a first buffer with an offset of NONE as
	  'from the middle of the stream', but only a first buffer
	  that has a valid buffer offset that's non-zero (see #345449).

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2006-09-14  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/icydemux/gsticydemux.c: (gst_icydemux_reset),
	(gst_icydemux_typefind_or_forward):
	* gst/icydemux/gsticydemux.h:
	  When we merge/collect multiple incoming buffers for typefinding
	  purposes, keep an initial 0 offset on the first outgoing buffer
	  as well (otherwise id3demux won't work right). Fixes #345449.
	  Also Make buffer metadata writable before setting buffer caps.

	* tests/check/elements/icydemux.c: (typefind_succeed),
	(cleanup_icydemux), (push_data), (GST_START_TEST),
	(icydemux_suite):
	  Small test case for the above.

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2006-09-13  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_peek_chunk),
	(gst_avi_demux_stream_index), (gst_avi_demux_sync),
	(gst_avi_demux_stream_header_push),
	(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
	(gst_avi_demux_loop):
	  More code reuse and better logging in _peek_chunk(). Reintroduce check
	  for chunk sizes before reading them (avoid oom). Better handling for 
	  invalid chunksizes when streaming.

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2006-09-11  Stefan Kost  <ensonic@users.sf.net>

	* gst/level/gstlevel.c: (gst_level_set_property):
	* gst/level/gstlevel.h:
          Fix type mixup in level->interval (gdouble<->guint64). Spotted by
          René Stadler

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2006-09-06  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes_pull),
	(gst_avi_demux_sync), (gst_avi_demux_stream_header_push),
	(gst_avi_demux_stream_data):
	  Revert one change to fix streaming avi (adapter size != data size).

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2006-09-04  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Frédéric Riss  <frederic.riss at gmail dot com>

	* gst/matroska/matroska-demux.c: (gst_matroska_track_free),
	(gst_matroska_demux_reset),
	(gst_matroska_demux_read_track_encodings),
	(gst_matroska_demux_add_stream), (gst_matroska_decode_buffer),
	(gst_matroska_demux_parse_blockgroup_or_simpleblock),
	(gst_matroska_demux_subtitle_caps):
	* gst/matroska/matroska-ids.h:
	  Add support for VOBSUB subtitle tracks and zlib-compressed
	  tracks. Make sure we start on a keyframe after a seek. (#343348)

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2006-09-04  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/matroska/matroska-demux.c: (gst_matroska_demux_push_hdr_buf),
	(gst_matroska_demux_push_flac_codec_priv_data),
	(gst_matroska_demux_push_xiph_codec_priv_data),
	(gst_matroska_demux_parse_blockgroup_or_simpleblock),
	(gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps):
	* gst/matroska/matroska-ids.h:
	  Add basic FLAC support (#311586), not perfect yet though, needs some
	  tweaking in flacdec; also, seeking could be better.
	  Do better bounds checking when deserialising vorbis stream headers
	  to make sure we don't read beyond the end of the buffer on bad input.

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2006-09-04  Wim Taymans  <wim@fluendo.com>

	Patch by: Alessandro Decina <alessandro at nnva dot org>

	* ext/annodex/gstcmmldec.c: (gst_cmml_dec_chain):
	Seeking back in a file containing a CMML stream errors out if the seek
	goes back up to the CMML headers. This is because after the seek the xml
	processing instruction <?xml ...?> is submitted to the xml parser again, 
	which results in an error. The attached patch fixes the problem. 
	Fixes #353908.

	* ext/annodex/gstcmmlenc.h:
	Fix authors name.


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2006-08-28  Andy Wingo  <wingo@pobox.com>

	* ext/raw1394/gstdv1394src.c (gst_dv1394src_from_raw1394handle):
	New helper function to lessen the ifdefs.
	(GST_INFO_OBJECT): 
	(gst_dv1394src_iso_receive): Use it.
	(gst_dv1394src_create): Also use the control sockets in iec61883
	mode.
	(gst_dv1394src_start, gst_dv1394src_stop): Always use a separate
	handle for AVC operations; fixes #348233.

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