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2007-08-16  Sebastian Dröge  <slomo@circular-chaos.org>

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

	* gst/audiofx/Makefile.am:
	* gst/audiofx/audiochebyshevfreqband.c:
	(gst_audio_chebyshev_freq_band_mode_get_type),
	(gst_audio_chebyshev_freq_band_base_init),
	(gst_audio_chebyshev_freq_band_dispose),
	(gst_audio_chebyshev_freq_band_class_init),
	(gst_audio_chebyshev_freq_band_init),
	(generate_biquad_coefficients), (calculate_gain),
	(generate_coefficients),
	(gst_audio_chebyshev_freq_band_set_property),
	(gst_audio_chebyshev_freq_band_get_property),
	(gst_audio_chebyshev_freq_band_setup), (process), (process_64),
	(process_32), (gst_audio_chebyshev_freq_band_transform_ip),
	(gst_audio_chebyshev_freq_band_start):
	* gst/audiofx/audiochebyshevfreqband.h:
	* gst/audiofx/audiochebyshevfreqlimit.c:
	(gst_audio_chebyshev_freq_limit_mode_get_type),
	(gst_audio_chebyshev_freq_limit_base_init),
	(gst_audio_chebyshev_freq_limit_dispose),
	(gst_audio_chebyshev_freq_limit_class_init),
	(gst_audio_chebyshev_freq_limit_init),
	(generate_biquad_coefficients), (calculate_gain),
	(generate_coefficients),
	(gst_audio_chebyshev_freq_limit_set_property),
	(gst_audio_chebyshev_freq_limit_get_property),
	(gst_audio_chebyshev_freq_limit_setup), (process), (process_64),
	(process_32), (gst_audio_chebyshev_freq_limit_transform_ip),
	(gst_audio_chebyshev_freq_limit_start):
	* gst/audiofx/audiochebyshevfreqlimit.h:
	* gst/audiofx/audiofx.c: (plugin_init):
	Add Chebyshev lowpass/highpass and bandpass/bandreject elements.
	Fixes #464800.

	* tests/check/Makefile.am:
	* tests/check/elements/.cvsignore:
	* tests/check/elements/audiochebyshevfreqband.c:
	(setup_audiochebyshevfreqband), (cleanup_audiochebyshevfreqband),
	(GST_START_TEST), (audiochebyshevfreqband_suite), (main):
	* tests/check/elements/audiochebyshevfreqlimit.c:
	(setup_audiochebyshevfreqlimit), (cleanup_audiochebyshevfreqlimit),
	(GST_START_TEST), (audiochebyshevfreqlimit_suite), (main):
	Add unit tests for the chebyshev filters.

	* 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-1394.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-shout2send.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	And add docs for the chebyshev filters. While doing
	that also run make update in docs/plugins.

Stefan Kost's avatar
Stefan Kost committed
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2007-08-16  Stefan Kost  <ensonic@users.sf.net>

	* ext/annodex/gstcmmltag.c:
	* gst/rtp/gstrtpvorbispay.c:
	  Make ro memory to share.

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2007-08-16  Wim Taymans  <wim.taymans@gmail.com>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
	Improve UDP performance by avoiding a select() when we have data
	available immediatly.

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2007-08-16  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_VOID__UINT_UINT),
	(gst_rtp_dec_class_init):
	* gst/rtsp/gstrtpdec.h:
	Add (dummy) SSRC management signals.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
	(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
	(find_stream), (gst_rtspsrc_create_stream), (new_session_pad),
	(request_pt_map), (gst_rtspsrc_do_stream_eos), (on_bye_ssrc),
	(on_timeout), (gst_rtspsrc_stream_configure_manager),
	(gst_rtspsrc_stream_push_event), (gst_rtspsrc_push_event),
	(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo),
	(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	* gst/rtsp/gstrtspsrc.h:
	Add connection-speed property.
	Add find_stream helper functions.
	Handle stream EOS based on BYE messages or SSRC timeout.
	Returns SUCCESS from the state change function as we hide our async
	elements from the parent.

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2007-08-16  Stefan Kost  <ensonic@users.sf.net>

	* gst/debug/rndbuffersize.c:
	  Fix da leak.

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2007-08-14  Stefan Kost  <ensonic@users.sf.net>

	* gst/debug/Makefile.am:
	* gst/debug/breakmydata.c:
	* gst/debug/gstdebug.c:
	* gst/debug/negotiation.c:
	* gst/debug/progressreport.c:
	* gst/debug/rndbuffersize.c:
	* gst/debug/testplugin.c:
	  Add new test element and clean-up the others a little.

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2007-08-12  Wim Taymans  <wim.taymans@gmail.com>

	* gst/qtdemux/qtdemux.c: (qtdemux_parse_node):
	Fix parsing of mp4a version 0 atoms. Fixes #465774.

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2007-08-10  Stefan Kost  <ensonic@users.sf.net>

	* gst/rtp/gstrtpilbcdepay.c:
	  Include stdlib.

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2007-08-10  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtp/gstrtpmpvdepay.c:
	Set the mpegversion in the caps so that autoplugging does not get
	confused.

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2007-08-09  Thomas Vander Stichele  <thomas at apestaart dot org>

	* po/hu.po:
	* po/uk.po:
	* po/vi.po:
	  Updated translations.

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2007-08-08  Michael Smith <msmith@fluendo.com>

	* gst/videobox/gstvideobox.c: (gst_video_box_ayuv_i420):
	  Render right border in the correct location.

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2007-08-08  Wim Taymans  <wim.taymans@gmail.com>

	Patch by: Olivier Crete <tester at tester dot ca>

	* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_setcaps):
	* gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_setcaps):
	Make mode property a string. Fixes #464475.

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2007-08-05  Stefan Kost  <ensonic@users.sf.net>

	* ext/flac/gstflacenc.c:
	  Widen caps to match decoder a bit and add more FIXMEs.

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2007-08-05  Stefan Kost  <ensonic@users.sf.net>

	patch by: Mark Nauwelaerts <manauw@skynet.be>

	* gst/avi/gstavimux.c:
	  Fix ODML index tag numbering. Fixes #463624.

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2007-08-03  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (get_default_rate_for_pt),
	(gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_stream_configure_tcp),
	(gst_rtspsrc_stream_configure_udp_sink):
	Fix default clock-rate for realmedia.
	Fix parsing of transport.
	Don't try to link NULL pads.

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2007-07-30  Tim-Philipp Müller  <tim at centricular dot net>

	* po/POTFILES.skip:
	  Add POTFILES.skip with list of source files that aren't disted at the
	  moment but contain translatable strings. Should hopefully pacify
	  broken tools and make it clearer that these files are left out
	  intentionally (#461600).

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2007-07-30  Edward Hervey  <bilboed@bilboed.com>

	* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_movie):
	If the buffer was entirely clipped ... don't try sending it :)

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2007-07-27  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams),
	(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_methods),
	(gst_rtspsrc_create_transports_string),
	(gst_rtspsrc_prepare_transports):
	If we don't hav a session manager, set the caps on outgoing buffers
	ourselves.
	Force PAUSE/PLAY methods for now until the extensions can overwrite.
	Append final bit of the transport string even when it does not contain a
	placeholder.

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2007-07-27  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_free),
	(gst_rtsp_ext_list_connect):
	* gst/rtsp/gstrtspext.h:
	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	(gst_rtspsrc_finalize), (gst_rtspsrc_send_cb):
	Clean up the interface list.
	Allow connecting to interface signals for the extensions.
	Remove old extension code.
	Free list on cleanup.
	Allow extensions to send additional RTSP messages.

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2007-07-27  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/gconf/gconf.c: (gst_gconf_render_bin_with_default):
	Handle a NULL gconf key gracefully by rendering the default element.

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2007-07-27  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspext.h:
	Fix include path for extension interface.

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2007-07-26  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audiofx/audioamplify.h:
	Also remove a now unecessary variable here.

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2007-07-26  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audiofx/audioamplify.c: (gst_audio_amplify_init),
	(gst_audio_amplify_setup), (gst_audio_amplify_transform_ip):
	* gst/audiofx/audiodynamic.c:
	(gst_audio_dynamic_set_process_function), (gst_audio_dynamic_init),
	(gst_audio_dynamic_setup), (gst_audio_dynamic_transform_ip):
	* gst/audiofx/audiodynamic.h:
	* gst/audiofx/audioinvert.c: (gst_audio_invert_init),
	(gst_audio_invert_setup), (gst_audio_invert_transform_ip):
	* gst/audiofx/audioinvert.h:
	Don't save format information ourselves, this is already saved in
	GstAudioFilter.

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2007-07-26  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_filter),
	(gst_rtsp_ext_list_stream_select):
	* gst/rtsp/gstrtspext.h:
	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams):
	Use rank to filter out extensions.
	Add url to stream_select interface call.

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2007-07-25  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/Makefile.am:
	* gst/rtsp/base64.c:
	* gst/rtsp/base64.h:
	* gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_filter),
	(gst_rtsp_ext_list_init), (gst_rtsp_ext_list_get),
	(gst_rtsp_ext_list_detect_server), (gst_rtsp_ext_list_before_send),
	(gst_rtsp_ext_list_after_send), (gst_rtsp_ext_list_parse_sdp),
	(gst_rtsp_ext_list_setup_media),
	(gst_rtsp_ext_list_configure_stream),
	(gst_rtsp_ext_list_get_transports),
	(gst_rtsp_ext_list_stream_select):
	* gst/rtsp/gstrtspext.h:
	* gst/rtsp/gstrtspsrc.c: (gst_rtsp_lower_trans_get_type),
	(gst_rtspsrc_class_init), (gst_rtspsrc_init),
	(gst_rtspsrc_finalize), (gst_rtspsrc_create_stream),
	(gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_flush), (gst_rtspsrc_do_seek),
	(gst_rtspsrc_sink_chain), (gst_rtspsrc_stream_configure_manager),
	(gst_rtspsrc_stream_configure_tcp),
	(gst_rtspsrc_stream_configure_mcast),
	(gst_rtspsrc_stream_configure_udp),
	(gst_rtspsrc_stream_configure_udp_sink),
	(gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_handle_request), (gst_rtspsrc_send_keep_alive),
	(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
	(gst_rtspsrc_loop_send_cmd), (gst_rtsp_auth_method_to_string),
	(gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth),
	(gst_rtspsrc_try_send), (gst_rtspsrc_send),
	(gst_rtspsrc_parse_methods),
	(gst_rtspsrc_create_transports_string),
	(gst_rtspsrc_prepare_transports), (gst_rtspsrc_setup_streams),
	(gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_close),
	(gst_rtspsrc_play), (gst_rtspsrc_pause),
	(gst_rtspsrc_change_state), (gst_rtspsrc_uri_set_uri):
	* gst/rtsp/gstrtspsrc.h:
	* gst/rtsp/rtsp.h:
	* gst/rtsp/rtspconnection.c:
	* gst/rtsp/rtspconnection.h:
	* gst/rtsp/rtspdefs.c:
	* gst/rtsp/rtspdefs.h:
	* gst/rtsp/rtspext.h:
	* gst/rtsp/rtspextwms.c:
	* gst/rtsp/rtspextwms.h:
	* gst/rtsp/rtspmessage.c:
	* gst/rtsp/rtspmessage.h:
	* gst/rtsp/rtsprange.c:
	* gst/rtsp/rtsprange.h:
	* gst/rtsp/rtsptransport.c:
	* gst/rtsp/rtsptransport.h:
	* gst/rtsp/rtspurl.c:
	* gst/rtsp/rtspurl.h:
	* gst/rtsp/sdp.h:
	* gst/rtsp/sdpmessage.c:
	* gst/rtsp/sdpmessage.h:
	* gst/rtsp/test.c:
	Use shiny new RTSP and SDP library.
	Implement RTSP extensions using the new interface.
	Remove a lot of old code.

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2007-07-24  Edward Hervey  <bilboed@bilboed.com>

	* gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
	Add codec mapping for '2vuy' (Raw YUV produced by FCP) and 'divx'.

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2007-07-24  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
	Don't unref the outgoing buffer twice when dropping it because it's
	outside of the segment.

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2007-07-24  Sebastian Dröge  <slomo@circular-chaos.org>

	* configure.ac:
	* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
	(gst_wavpack_dec_chain), (gst_wavpack_dec_sink_event):
	Use the new buffer clipping function from gstaudio here and
	require gst-plugins-base CVS.
	* tests/check/elements/wavpackdec.c: (GST_START_TEST):
	For framed Wavpack buffers we require a valid timestamp.

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2007-07-23  Wim Taymans  <wim.taymans@gmail.com>

	* gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment),
	(gst_qtdemux_clip_buffer), (gst_qtdemux_loop_state_movie),
	(qtdemux_parse_trak), (qtdemux_video_caps), (qtdemux_audio_caps):
	Clip raw audio and video when we can, keep track of current output
	segment.
	Don't leak buffers and events when there is no output pad.
	Improve debugging here and there.

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2007-07-23  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	  Sync liboil check with plugins-base.

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2007-07-20  Stefan Kost  <ensonic@users.sf.net>

	* ext/annodex/Makefile.am:
	  Fix CFLAGS/LIBS.

	* ext/cdio/gstcdiocddasrc.c:
	* ext/libpng/gstpngdec.c: (gst_pngdec_task):
	  Include stdlib

	* ext/cairo/Makefile.am:
	* gst/videofilter/Makefile.am:
	* tests/examples/level/Makefile.am:
	  Use $(LIBM) instead of -lm

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2007-07-18  Stefan Kost  <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2src.c:
	  Add another example pipeline.

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2007-07-18  Stefan Kost  <ensonic@users.sf.net>

	Patch by: Alexander Eichner <alexeichi@yahoo.de>

	* sys/v4l2/gstv4l2src.c: (gst_v4l2src_init):
	  Use define here.

	* sys/v4l2/gstv4l2tuner.c:
	(gst_v4l2_tuner_set_frequency_and_notify):
	  Don't touch the property - its still disabled.

	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_probe_caps_for_format),
	(gst_v4l2src_grab_frame), (gst_v4l2src_get_size_limits):
	* sys/v4l2/v4l2src_calls.h:
	  Improve fallback format negotionation. Fixes #451388

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2007-07-18  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/elements/videocrop.c: (GST_START_TEST):
	  Fix the test.

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2007-07-18  Stefan Kost  <ensonic@users.sf.net>

	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* ext/jpeg/gstjpegdec.c:
	* ext/libpng/gstpngdec.c: (gst_pngdec_task),
	(gst_pngdec_sink_setcaps):
	  More docs. More logs in pngdec.

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2007-07-17  Stefan Kost  <ensonic@users.sf.net>

	* sys/v4l2/gstv4l2src.c: (gst_v4l2src_init):
	  Initialize num_buffers with minimum value.

	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
	(gst_v4l2src_probe_caps_for_format), (gst_v4l2src_grab_frame):
	  Handle frame-size query failure gracefully.

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2007-07-16  Wim Taymans  <wim.taymans@gmail.com>

	* gst/qtdemux/qtdemux.c: (qtdemux_parse_node):
	Fix parsing of esds atoms inside mp4a atoms so that we can set correct
	codec_info for AAC audio. Fixes #457097 along with a whole other bunch
	of qt/aac files.

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2007-07-16  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/wavpack/gstwavpackdec.c:
	(gst_wavpack_dec_clip_outgoing_buffer):
	Fix buffer clipping to correctly clip to the segment stop.

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2007-07-13  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	* tests/Makefile.am:
	Remove bogus check for libcheck, since we check for
	gstreamer-check and it pulls in the required info from there,
	and we weren't actually _using_ the information for libcheck
	ourselves anyway.

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2007-07-12  Stefan Kost  <ensonic@users.sf.net>

	* configure.ac:
	  Use pkg-config to locate check.

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2007-07-11  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/cairo/gsttimeoverlay.c: (gst_cairo_time_overlay_transform):
	* ext/jpeg/gstjpegenc.c: (gst_jpegenc_chain):
	* ext/libpng/gstpngenc.c: (gst_pngenc_chain):
	* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
	* gst/debug/gstnavigationtest.c: (gst_navigationtest_transform):
	* gst/effectv/gstaging.c: (gst_agingtv_transform):
	* gst/effectv/gstdice.c: (gst_dicetv_transform):
	* gst/effectv/gstedge.c: (gst_edgetv_transform):
	* gst/effectv/gstquark.c: (gst_quarktv_transform):
	* gst/effectv/gstrev.c: (gst_revtv_transform):
	* gst/effectv/gstshagadelic.c: (gst_shagadelictv_transform):
	* gst/effectv/gstvertigo.c: (gst_vertigotv_transform):
	* gst/effectv/gstwarp.c: (gst_warptv_transform):
	* gst/matroska/matroska-demux.c:
	(gst_matroska_demux_add_wvpk_header),
	(gst_matroska_demux_check_subtitle_buffer),
	(gst_matroska_decode_buffer):
	* gst/videofilter/gstvideoflip.c: (gst_video_flip_transform):
	  Fix build against core CVS.

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2007-07-10  Edward Hervey  <bilboed@gmail.com>

	* gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
	Don't return GST_FLOW_ERROR when pushing an event returns FALSE. We
	don't have enough granularity to convert that boolean into a
	GstFlowReturn.

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2007-07-06  Michael Smith <msmith@fluendo.com>

	* gst/law/alaw-decode.c: (alawdec_sink_setcaps),
	(gst_alawdec_class_init), (gst_alawdec_init), (gst_alawdec_chain),
	(gst_alawdec_change_state):
	* gst/law/alaw-decode.h:
	* gst/law/mulaw-decode.c: (mulawdec_sink_setcaps),
	(gst_mulawdec_class_init), (gst_mulawdec_init),
	(gst_mulawdec_chain), (gst_mulawdec_change_state):
	* gst/law/mulaw-decode.h:
	  Fix capsnego bogosity in *law decoders. 

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2007-07-06  Michael Smith <msmith@fluendo.com>

	* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_init),
	(gst_smokeenc_setcaps), (gst_smokeenc_chain),
	(gst_smokeenc_change_state):
	* ext/jpeg/gstsmokeenc.h:
	  Remove stupidity in get/set caps functions.
	  Fix some refcounting problems.

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2007-07-06  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/libpng/gstpngdec.c: (gst_pngdec_caps_create_and_set):
	Remove endianness-flipping hack that seems to have been required
	only because of a bug in ffmpegcolorspace.
	Partially Fixes: #451908

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2007-07-05  Stefan Kost  <ensonic@users.sf.net>

	* docs/plugins/Makefile.am:
	  Simplify --extra-dir as gtkdoc scans recursively.

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2007-07-03  Wim Taymans,,,  <set EMAIL_ADDRESS environment variable>

	Patch by: Tommi Myöhänen  <ext-tommi dot myohanen at nokia dot com>

	* gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_setcaps):
	Set the encoding-name in the rtp caps to all uppercase, as required by
	the caps spec.
	Some small cleanups in the error paths. Fixes #453037.

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2007-06-28  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/wavpack/gstwavpackparse.c:
	(gst_wavpack_parse_index_get_last_entry),
	(gst_wavpack_parse_index_get_entry_from_sample),
	(gst_wavpack_parse_index_append_entry), (gst_wavpack_parse_reset),
	(gst_wavpack_parse_scan_to_find_sample):
	* ext/wavpack/gstwavpackparse.h:
	Use a GSList for the GArray that is used like a list anyway.

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2007-06-28  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps),
	(gst_gdk_pixbuf_class_init), (gst_gdk_pixbuf_flush),
	(gst_gdk_pixbuf_sink_event), (gst_gdk_pixbuf_change_state):
	  Add state change function where we set 0/1 as default framerate in
	  case our setcaps function isn't called, like it might not in a
	  filesrc ! gdkpixbufdec scenario. Fixes assertion triggered by
	  gdkpixbufdec trying to create caps with a 0/0 framerate.
	  Also post an error message on the bus if gst_pad_push() fails when
	  called from our sink event handler (+1 for flow returns for event
	  functions in 0.11) instead of failing silently.

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2007-06-27  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_configure_caps):
	Cast stack args to the proper types. Fixes #451249.

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2007-06-27  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
	(new_session_pad), (gst_rtspsrc_setup_streams):
	* gst/rtsp/gstrtspsrc.h:
	For container formats we only need to activate one of the streams so
	that we correctly signal no-more-pads. Fixes #451015.

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2007-06-25  Stefan Kost  <ensonic@users.sf.net>

	* docs/plugins/gst-plugins-good-plugins.args:
	* 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-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-ladspa.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-quicktime.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtsp.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-videocrop.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 docs with caps info.

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2007-06-25  Tim-Philipp Müller  <tim at centricular dot net>

	* po/POTFILES.in:
	  Add more files with translatable strings (#450878).

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2007-06-22  Jan Schmidt  <thaytan@noraisin.net>

	* MAINTAINERS:
	Updating all the maintainers files

Edward Hervey's avatar
Edward Hervey committed
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2007-06-22  Edward Hervey  <edward@fluendo.com>

	* ext/flac/gstflactag.c: (gst_flac_tag_init):
	* gst/interleave/deinterleave.c: (deinterleave_init),
	(deinterleave_sink_link):
	* gst/interleave/interleave.c: (interleave_init):
	* gst/median/gstmedian.c: (gst_median_init):
	* gst/oldcore/gstmultifilesrc.c: (gst_multifilesrc_init):
	Fix memory leaks.
	* tests/check/elements/id3demux.c: (pad_added_cb):
	Remove unused variable.

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2007-06-21  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/gconf/gconf.h:
	Make the prototype of gst_gconf_get_key_for_sink_profile
	match the implementation.
	Patch by: Damien Carbery <damien dot carbery at sun dot com>
	Fixes: #449747

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2007-06-20  Michael Smith <msmith@fluendo.com>

	* gst/rtp/gstrtpdepay.c:
	  Fix description - rtpdepay is not a payloader.

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2007-06-20  Stefan Kost  <ensonic@users.sf.net>

	* gst/qtdemux/qtdemux.c: (qtdemux_parse_samples),
	(qtdemux_video_caps):
	* gst/qtdemux/qtdemux_fourcc.h:
	  Add MJPG to the variants of motion jpeg.

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2007-06-19  Tim-Philipp Müller  <tim at centricular dot net>

	* tests/check/Makefile.am:
	* tests/check/elements/audiopanorama.c: (GST_START_TEST):
	* tests/check/elements/videocrop.c: (GST_START_TEST):
	* tests/check/elements/videofilter.c:
	* tests/check/elements/wavpackdec.c: (GST_START_TEST):
	* tests/check/elements/wavpackparse.c: (GST_START_TEST):
	  Add GST_OPTION_CFLAGS to CFLAGS when building unit tests, so the
	  error flags are included and it errors out on compiler warnings
	  for CVS builds; remove unused variables in various unit tests.

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2007-06-19  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
	(rtsp_connection_close), (rtsp_connection_free):
	Use threadsafe inet_ntop to convert an ip number to a string. 
	Fixes #447961.
	Don't leak fd (and ip) when freeing a connection without first closing
	it.

Jan Schmidt's avatar
Jan Schmidt committed
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2007-06-19  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	Back to CVS

	* gst-plugins-good.doap:
	Add 0.10.6 to the doap file.

Jan Schmidt's avatar
Jan Schmidt committed
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=== release 0.10.6 ===

2007-06-18  Jan Schmidt <thaytan@mad.scientist.com>

	* configure.ac:
	  releasing 0.10.6, "Wobble Board"

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2007-06-17  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
	(rtsp_connection_free):
	  Revert previous commit again, since we are frozen (sorry).

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2007-06-17  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Peter Kjellerstedt <pkj at axis com>

	* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
	(rtsp_connection_free):
	  inet_ntoa() uses a static buffer internally, so we need to copy the
	  returned string if we want to store it for later (#447961).

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2007-06-15  Jan Schmidt  <thaytan@mad.scientist.com>

	* win32/vs6/autogen.dsp:
	* win32/vs6/gst_plugins_good.dsw:
	* win32/vs6/libgstalaw.dsp:
	* win32/vs6/libgstalpha.dsp:
	* win32/vs6/libgstalphacolor.dsp:
	* win32/vs6/libgstapetag.dsp:
	* win32/vs6/libgstaudiofx.dsp:
	* win32/vs6/libgstauparse.dsp:
	* win32/vs6/libgstautodetect.dsp:
	* win32/vs6/libgstavi.dsp:
	* win32/vs6/libgstcutter.dsp:
	* win32/vs6/libgstdirectdraw.dsp:
	* win32/vs6/libgstdirectsound.dsp:
	* win32/vs6/libgsteffectv.dsp:
	* win32/vs6/libgstflx.dsp:
	* win32/vs6/libgstgoom.dsp:
	* win32/vs6/libgsticydemux.dsp:
	* win32/vs6/libgstid3demux.dsp:
	* win32/vs6/libgstinterleave.dsp:
	* win32/vs6/libgstjpeg.dsp:
	* win32/vs6/libgstlevel.dsp:
	* win32/vs6/libgstmatroska.dsp:
	* win32/vs6/libgstmedian.dsp:
	* win32/vs6/libgstmonoscope.dsp:
	* win32/vs6/libgstmulaw.dsp:
	* win32/vs6/libgstmultipart.dsp:
	* win32/vs6/libgstqtdemux.dsp:
	* win32/vs6/libgstrtp.dsp:
	* win32/vs6/libgstrtsp.dsp:
	* win32/vs6/libgstsmpte.dsp:
	* win32/vs6/libgstspeex.dsp:
	* win32/vs6/libgstudp.dsp:
	* win32/vs6/libgstvideobalance.dsp:
	* win32/vs6/libgstvideobox.dsp:
	* win32/vs6/libgstvideocrop.dsp:
	* win32/vs6/libgstvideoflip.dsp:
	* win32/vs6/libgstvideomixer.dsp:
	* win32/vs6/libgstwaveform.dsp:
	* win32/vs6/libgstwavenc.dsp:
	* win32/vs6/libgstwavparse.dsp:
	Mark *.dsp & *.dsw as binary files and convert to DOS line
	endings, as they don't load into VS6 correctly otherwise.

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2007-06-15  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	(rtsp_connection_connect):
	Fix the MingW build. 
	Patch By: Vincent Torri <vtorri at univ-evry dot fr>
	Fixes: #446981

Jan Schmidt's avatar
Jan Schmidt committed
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2007-06-14  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/elements/.cvsignore:
	* tests/icles/.cvsignore:
	Hush the buildbots up

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2007-06-14  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	* sys/Makefile.am:
	* sys/directdraw/Makefile.am:
	* sys/directsound/Makefile.am:
	* sys/waveform/Makefile.am:
	Make sure to dist everything needed for win32 builds.

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2007-06-14  Edward Hervey  <edward@fluendo.com>

	* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
	For AMR-NB streams, export the AMRSpecificBox as codec_data on the
	caps.
	Fixes #447458

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2007-06-13  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps):
	Make sure we allocate enough memory for the codec_data.
	Fixes #447210.

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2007-06-12  Sebastien Moutte  <sebastien@moutte.net>

	* win32/MANIFEST:
	Add videocrop project file to the win32 manifest.
	* win32/vs6/gst_plugins_good.dsw:
	Add qtdemux,videocrop and waveform projects to the workspace.
	* win32/vs6/libgstqtdemux.dsp:
	Add zlib to the link list of qtdemux.
	* win32/vs6/libgstvideocrop.dsp:
	Add a project file for videocrop.

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2007-06-12  Jan Schmidt  <thaytan@mad.scientist.com>

	* po/POTFILES.in:
	Add qtdemux for translation

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2007-06-12  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* gst-plugins-good.spec.in:
	* sys/Makefile.am:
	* tests/check/Makefile.am:
	* tests/icles/Makefile.am:
	* tests/icles/videocrop-test.c:
	Move videocrop and osxvideo from -bad.

Jan Schmidt's avatar
Jan Schmidt committed
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2007-06-12  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	* 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-qtdemux.xml:
	* docs/plugins/inspect/plugin-quicktime.xml:
	* win32/MANIFEST:
	Move qtdemux from -bad.

	* gst-plugins-good.spec.in:
	Update spec file to reflect moving of qtdemux and wavpack

Jan Schmidt's avatar
Jan Schmidt committed
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2007-06-12  Jan Schmidt  <thaytan@mad.scientist.com>
	
816
	* win32/MANIFEST:
Jan Schmidt's avatar
Jan Schmidt committed
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	* docs/plugins/Makefile.am:
	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/inspect/plugin-directdraw.xml:
	* docs/plugins/inspect/plugin-directsound.xml:
	* docs/plugins/inspect/plugin-waveform.xml:
	Move the waveform plugin from -bad too. Update the inspect xml
	files to mention Plugins Good instead of Plugins Bad.

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2007-06-12  Andy Wingo  <wingo@pobox.com>

	* sys/v4l2/v4l2src_calls.c (gst_v4l2_buffer_finalize)
	(gst_v4l2_buffer_class_init, gst_v4l2_buffer_get_type)
	(gst_v4l2_buffer_new): Behave more like ximagesink's buffers, with
	finalization and resuscitation. No longer public.
	(gst_v4l2_buffer_pool_finalize, gst_v4l2_buffer_pool_init)
	(gst_v4l2_buffer_pool_class_init, gst_v4l2_buffer_pool_get_type)
	(gst_v4l2_buffer_pool_new, gst_v4l2_buffer_pool_activate)
	(gst_v4l2_buffer_pool_destroy): Make the pool follow common
	miniobject semantics, and be threadsafe.
	(gst_v4l2src_queue_frame): Remove this function, as we just call
	the ioctls directly in the two places where we queue buffers.
	(gst_v4l2src_grab_frame): Return a flowreturn and fill the buffer
	directly.
	(gst_v4l2src_capture_init): Use the new buffer_pool_new function
	to allocate the pool, which also preallocates the GstBuffers.
	(gst_v4l2src_capture_start): Call buffer_pool_activate instead of
	queueing the frames directly.
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	(gst_v4l2src_grab_frame): Return a copy of the pool buffer if all
	mmap buffers have been dequeued.
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	* sys/v4l2/gstv4l2src.h (struct _GstV4l2BufferPool): Make this a
	real MiniObject instead of rolling our own refcounting and
	finalizing. Give it a lock.
	(struct _GstV4l2Buffer): Remove one intermediary object, having
	the buffers hold the struct v4l2_buffer directly.

	* sys/v4l2/gstv4l2src.c (gst_v4l2src_set_caps): Pass the caps to
	capture_init so that it can set them on the buffers that it will
	create.
	(gst_v4l2src_get_read): For better or for worse, include the
	timestamping and offsetting code here; really we should be using
	bufferalloc though.
	(gst_v4l2src_get_mmap): Just make grab_frame return one of our
	preallocated, mmap'd buffers.

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2007-06-11  Wim Taymans  <wim@fluendo.com>

	Patch by: daniel fischer <dan at f3c dot com>

	* sys/ximage/gstximagesrc.c: (gst_ximage_src_start),
	(gst_ximage_src_get_caps):
	Actually use the display_name property so that we can dump any
	available X display. Fixes #445905.

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2007-06-11  Wim Taymans  <wim@fluendo.com>

	Patch by: Tommi Myöhänen  <ext-tommi dot myohanen at nokia dot com>

	* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_setcaps):
	* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_setcaps):
	Add missing rate fields to caps. Fixes #441118.

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2007-06-10  Sebastien Moutte  <sebastien@moutte.net>

	* win32/vs6/gst_plugins_good.dsw:
	* win32/vs8/gst-plugins-good.sln:
	Add DirectSound and DirectDraw sinks project files to
	workspace and solution files.

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2007-06-10  Sebastian Dröge  <slomo@circular-chaos.org>

	Patch by: Josh Coalson <xflac at yahoo dot com>,
	updated by Alexis Ballier <aballier at gentoo dot org>:

	* configure.ac:
	* ext/flac/gstflacdec.c: (gst_flac_dec_reset_decoders),
	(gst_flac_dec_setup_seekable_decoder),
	(gst_flac_dec_setup_stream_decoder), (gst_flac_dec_seek),
	(gst_flac_dec_tell), (gst_flac_dec_length), (gst_flac_dec_eof),
	(gst_flac_dec_read_seekable), (gst_flac_dec_read_stream):
	* ext/flac/gstflacdec.h:
	* ext/flac/gstflacenc.c: (gst_flac_enc_init),
	(gst_flac_enc_finalize), (gst_flac_enc_set_metadata),
	(gst_flac_enc_sink_setcaps), (gst_flac_enc_update_quality),
	(gst_flac_enc_seek_callback), (gst_flac_enc_write_callback),
	(gst_flac_enc_tell_callback), (gst_flac_enc_sink_event),
	(gst_flac_enc_chain), (gst_flac_enc_set_property),
	(gst_flac_enc_get_property), (gst_flac_enc_change_state):
	* ext/flac/gstflacenc.h:
	Add support for flac >= 1.1.3 which changed the API. Fixes bug #385887.
	
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2007-06-09  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_sink_set_caps):
	Remove workaround for bug #421543. This is fixed in core 0.10.13 and
	not necessary anymore as we need at least that core version. 

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2007-06-09  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
	(gst_wavpack_dec_chain):
	* ext/wavpack/gstwavpackdec.h:
	* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_reset),
	(gst_wavpack_parse_push_buffer):
	* ext/wavpack/gstwavpackparse.h:
	Improve discont handling by checking if the next Wavpack block has
	the expected, following block index.

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2007-06-08  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/rtp/gstrtpmp4vpay.c (gst_rtp_mp4vpay_details):
	  Fix element description.

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2007-06-08  Thomas Vander Stichele  <thomas at apestaart dot org>

	* configure.ac:
	* 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/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* docs/plugins/inspect/plugin-autodetect.xml:
	* docs/plugins/inspect/plugin-gconfelements.xml:
	* docs/plugins/inspect/plugin-ladspa.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-wavpack.xml:
	* ext/Makefile.am:
	* tests/check/Makefile.am:
	  move wavpack plugin.  See #352605.

949 950 951 952 953 954 955 956 957 958 959
2007-06-08  Jan Schmidt  <thaytan@mad.scientist.com>

	* configure.ac:
	* 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:
	* sys/Makefile.am:
	* win32/MANIFEST:
	Add DirectDraw & DirectSound plugins to the build and docs.

960 961 962 963 964 965
2007-06-08  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/dv/gstdvdemux.c: (gst_dvdemux_loop):
	* ext/libpng/gstpngdec.c: (user_read_data), (gst_pngdec_task):
	  When operating in pull mode, error out correct on not-linked.

966 967 968 969 970 971 972
2007-06-06  Andy Wingo  <wingo@pobox.com>

	* sys/v4l2/v4l2src_calls.c (gst_v4l2src_probe_caps_for_format)
	(gst_v4l2src_probe_caps_for_format_and_size): Only probe for
	format and size if the ioctls are defined; should fix compilation
	on Linux < 2.16.19.

973 974 975 976 977 978 979 980
2007-06-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/videobox/gstvideobox.c: (gst_video_box_ayuv_i420):
	  Printf fixes in debug statements; use LOG level for debug statements
	  that are printed for each and every frame; convert c++ comments to
	  C-style comments; not much point using g_try_malloc() if we then not
	  even check the return value.

981 982 983 984 985 986 987 988 989
2007-06-05  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  Bump requirements to released versions (core and base 0.10.13).

	* gst/icydemux/gsticydemux.c: (gst_icydemux_unicodify):
	  Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
	  own implementation.

990 991 992 993 994 995 996 997 998 999 1000 1001 1002 1003 1004
2007-06-05  Andy Wingo  <wingo@pobox.com>

	* sys/v4l2/gstv4l2src.c (gst_v4l2src_start, gst_v4l2src_stop): Add
	some useless comments.

	* sys/v4l2/v4l2src_calls.c (gst_v4l2src_capture_init): Don't queue
	frames before calling STREAMON, that might leave them in a state
	where they can't be dequeued if we go back to NULL without calling
	STREAMON, according to the docs.
	(gst_v4l2src_capture_start): Enqueue buffers here instead, right
	before we call STREAMON.
	(gst_v4l2src_capture_deinit): Remove crack to work around dequeue
	failures. (For me this code hung.) The pool refcounting is still
	crack; added a note to that effect.

1005 1006 1007 1008 1009 1010 1011
2007-06-05  Wim Taymans  <wim@fluendo.com>

	* gst/multipart/multipartmux.c: (gst_multipart_mux_class_init),
	(gst_multipart_mux_get_mime), (gst_multipart_mux_collected):
	Add support for mapping gst structure names to the MIME type equivalent.
	Implemented for audio/x-mulaw->audio/basic. Fixes #442874.

1012 1013 1014 1015 1016 1017 1018 1019 1020
2007-06-03  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
	(gst_wavenc_sink_setcaps), (gst_wavenc_format_samples),
	(gst_wavenc_chain), (gst_wavenc_change_state):
	* gst/wavenc/gstwavenc.h:
	Properly write wav files with width!=depth by having the depth most
	significant bytes set and all others zero. Fixes #442535.

1021 1022 1023 1024 1025
2007-06-01  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/rtspconnection.c:
	Add include to make buildbot happy.

1026 1027 1028 1029 1030 1031 1032 1033 1034 1035 1036 1037 1038 1039 1040 1041 1042 1043 1044 1045 1046 1047 1048 1049 1050 1051 1052
2007-06-01  Wim Taymans  <wim@fluendo.com>

	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	(rtsp_connection_connect), (add_date_header),
	(rtsp_connection_send), (parse_response_status),
	(parse_request_line), (parse_line), (rtsp_connection_receive):
	* gst/rtsp/rtspdefs.c: (rtsp_version_as_text):
	* gst/rtsp/rtspdefs.h:
	* gst/rtsp/rtspmessage.c: (key_value_foreach),
	(rtsp_message_init_request), (rtsp_message_init_response),
	(rtsp_message_remove_header), (rtsp_message_append_headers),
	(rtsp_message_dump):
	* gst/rtsp/rtspmessage.h:
	Improves version checking, allowing an RTSP server to reply with "505
	RTSP Version not supported.
	Adds a Date header to all messages.
	Replies with RTSP_EPARSE rather than RTSP_EINVALID in cases where we
	want to be able to send a response even if something in the request was
	invalid. EINVAL is only used when passing wrong arguments to functions.
	Do not handle an invalid method in parse_request_line(). Defer this to
	the caller so it can respond with "405 Method Not Allowed".
	Improves parsing of the timeout parameter to the Session header,
	allowing whitespace after the semicolon. 
	Avoids a compiler warning due to variables shadowing a function argument.

1053 1054 1055 1056 1057 1058 1059 1060 1061 1062 1063 1064 1065 1066
2007-06-01  Wim Taymans  <wim@fluendo.com>

	Based on Patch by: Daniel Charles <dcharles at ti dot com>

	* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps),
	(gst_rtp_amr_depay_process):
	* gst/rtp/gstrtpamrdepay.h:
	* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_base_init),
	(gst_rtp_amr_pay_class_init), (gst_rtp_amr_pay_init),
	(gst_rtp_amr_pay_setcaps), (gst_rtp_amr_pay_handle_buffer):
	* gst/rtp/gstrtpamrpay.h:
	Add support for AMR-WB.
	Small cleanups such as using BOILERPLATE.

1067 1068 1069 1070 1071
2007-05-31  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_configure_stream):
	Fix compile warning when debug is disabled as spotted bu Saur on IRC.

1072 1073
2007-05-30  Andy Wingo  <wingo@pobox.com>

1074 1075 1076 1077
	* sys/v4l2/gstv4l2object.h: 
	* sys/v4l2/gstv4l2object.c (gst_v4l2_object_new): Revert some
	unintended changes.

1078 1079 1080 1081 1082 1083 1084 1085 1086 1087 1088 1089 1090 1091 1092 1093 1094 1095 1096 1097 1098 1099 1100 1101 1102 1103 1104 1105 1106 1107 1108 1109 1110 1111 1112 1113 1114 1115 1116 1117 1118 1119 1120 1121 1122 1123
	* sys/v4l2/v4l2src_calls.h: 
	* sys/v4l2/v4l2src_calls.c (gst_v4l2src_fill_format_list): Store
	the format list in the order that the driver gives it to us.
	(gst_v4l2src_probe_caps_for_format_and_size)
	(gst_v4l2src_probe_caps_for_format): New functions, fill GstCaps
	based on the capabilities of the device.
	(gst_v4l2src_grab_frame): Update for object variable renaming.
	(gst_v4l2src_set_capture): Update to be strict in its parameters,
	as in the set_caps below.
	(gst_v4l2src_capture_init): Update for object variable renaming,
	and reflow.
	(gst_v4l2src_capture_start, gst_v4l2src_capture_stop)
	(gst_v4l2src_capture_deinit): Update for object variable renaming.
	(gst_v4l2src_update_fps, gst_v4l2src_set_fps)
	(gst_v4l2src_get_fps): Remove; these functions don't have much
	meaning outside of an atomic set_caps method.
	(gst_v4l2src_buffer_new): Don't set buffer duration, it is not
	known.

	* sys/v4l2/gstv4l2tuner.c (gst_v4l2_tuner_set_channel): Remove
	call to update_fps; not sure about this change.
	(gst_v4l2_tuner_set_norm): Work around the fact that for the
	moment we don't have an update_fps_func.

	* sys/v4l2/gstv4l2src.h (struct _GstV4l2Src): Don't put v4l2
	structures in the object, just store what we need. Do store the
	probed caps of the device. Don't store the current frame rate.

	* sys/v4l2/gstv4l2src.c (gst_v4l2src_init): Remove the
	update_fps_function, for now. Update for new object variable
	naming.
	(gst_v4l2src_set_property, gst_v4l2src_get_property): Update for
	new object variable naming.
	(gst_v4l2src_v4l2fourcc_to_structure): Rename from ..._to_caps.
	(gst_v4l2_structure_to_v4l2fourcc): Rename from ...caps_to_....
	(gst_v4l2src_get_caps): Rework to probe the device for supported
	frame sizes and frame rates.
	(gst_v4l2src_set_caps): Rework to be strict in the given
	parameters: if someone asks us to have a certain size and rate,
	that is what we configure.
	(gst_v4l2src_get_read): Update for object variable naming. Don't
	leak buffers on short reads.
	(gst_v4l2src_get_mmap): Update for object variable naming, and add
	comments.
	(gst_v4l2src_create): Update for object variable naming.

1124 1125 1126 1127 1128 1129 1130 1131
2007-05-30  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_base_init),
	(gst_avi_demux_reset), (gst_avi_demux_parse_stream):
	* gst/avi/gstavidemux.h:
	  Parse subtitle text streams instead of erroring out (#442034). Still
	  needs a parser for the subtitles to actually show up.

1132 1133 1134 1135 1136 1137 1138 1139 1140
2007-05-30  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_push_event),
	(gst_avi_demux_loop):
	  Make _push_event() return TRUE if the event could be pushed on at
	  least one pad and not only if it could be pushed on all pads,
	  otherwise we'll end up posting an error message on EOS if one or
	  more source pads are not connected.

1141 1142 1143 1144 1145
2007-05-28  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/rtsptransport.c:
	Use renamed RTP bin.

1146 1147 1148 1149 1150 1151 1152 1153 1154 1155 1156 1157 1158 1159 1160 1161
2007-05-28  Wim Taymans  <wim@fluendo.com>

	Based on patch by: Dejan Sakelšak <sakdean at gmail dot com>

	* gst/videobox/gstvideobox.c: (gst_video_box_class_init),
	(gst_video_box_set_property), (gst_video_box_transform_caps),
	(video_box_recalc_transform), (gst_video_box_set_caps),
	(gst_video_box_get_unit_size), (gst_video_box_apply_alpha),
	(gst_video_box_ayuv_ayuv), (gst_video_box_clear), (UVfloor),
	(UVceil), (gst_video_box_ayuv_i420), (gst_video_box_i420_ayuv),
	(gst_video_box_i420_i420), (gst_video_box_transform),
	(plugin_init):
	Add AYUV->AYUV and AYUV->I420 formats. 
	Fix negotiation and I420->AYUV conversion.
	Fixes #429329.

1162 1163 1164 1165 1166 1167 1168 1169
2007-05-26  Wim Taymans  <wim@fluendo.com>

	* ext/speex/gstspeexdec.c: (speex_dec_chain_parse_data):
	Use different variables for nested for loops so that the outer loop
	functions properly and speex files with multiple frames per buffer work
	properly.
	Fixes #441408.

1170 1171 1172 1173 1174
2007-05-25  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/id3demux/gstid3demux.c: (gst_id3demux_sink_event):
	  Don't leak newsegment events.

1175 1176 1177 1178 1179 1180
2007-05-25  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/wavparse/Makefile.am:
	  Add '-lm' to LIBS for ceil(), don't assume one of our dependencies
	  drags it in.

1181 1182 1183 1184 1185 1186 1187 1188 1189 1190 1191 1192 1193
2007-05-25  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/flac/gstflacenc.c: (gst_flac_enc_init),
	(notgst_value_array_append_buffer),
	(gst_flac_enc_process_stream_headers),
	(gst_flac_enc_write_callback), (gst_flac_enc_chain),
	(gst_flac_enc_change_state):
	* ext/flac/gstflacenc.h:
	  Collect headers, add "streamheader" field to output caps and set
	  BUFFER_IN_CAPS flag on pushed header buffers. That way oggmux
	  produces output according to the official FLAC-to-Ogg mapping
	  instead of completely broken files. Fixes #426044.

1194 1195 1196 1197 1198 1199 1200 1201 1202 1203 1204 1205 1206 1207 1208
2007-05-25  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/id3demux/gstid3demux.c: (gst_id3demux_reset),
	(gst_id3demux_send_new_segment), (gst_id3demux_chain),
	(gst_id3demux_sink_event):
	* gst/id3demux/gstid3demux.h:
	* gst/apetag/gsttagdemux.c: (gst_tag_demux_reset),
	(gst_tag_demux_chain), (gst_tag_demux_sink_event),
	(gst_tag_demux_send_new_segment):
	Handle and adjust new-segment events so that downstream really
	sees a stream with the tag pieces stripped off the front and back.
	Fixes strangeness in seeking when mp3 decoders use the new-segment
	byte position to estimate their current playback position timestamp
	and then the arriving buffers don't match up.

1209 1210 1211 1212 1213 1214
2007-05-25  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_detect):
	  Don't unnecessarily perform a READY->NULL->READY transition on the
	  detected audio sink when starting up. Fixes: #440127

1215 1216 1217 1218 1219 1220
2007-05-24  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/flac/gstflacenc.c: (gst_flac_enc_sink_setcaps),
	(gst_flac_enc_chain):
	  Don't crash in chain function if setcaps hasn't been called.

1221 1222 1223 1224 1225
2007-05-24  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_methods):
	Init value to avoid infinte loops.

1226 1227 1228 1229 1230 1231 1232 1233 1234 1235 1236 1237 1238 1239 1240 1241 1242 1243 1244 1245 1246 1247 1248 1249 1250
2007-05-24  Wim Taymans  <wim@fluendo.com>

	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_auth),
	(gst_rtspsrc_try_send), (gst_rtspsrc_parse_methods),
	(gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
	(gst_rtspsrc_play):
	(rtsp_connection_send), (rtsp_connection_receive):
	* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send):
	Fix for new API.

	* gst/rtsp/rtspconnection.c: (add_auth_header),
	Only add authorisation and session headers when sending messages.

	* gst/rtsp/rtspmessage.c: (key_value_foreach), (rtsp_message_init),
	(rtsp_message_init_request), (rtsp_message_init_response),
	(rtsp_message_unset), (rtsp_message_add_header),
	(rtsp_message_remove_header), (rtsp_message_get_header),
	(rtsp_message_append_headers), (dump_key_value),
	(rtsp_message_dump):
	* gst/rtsp/rtspmessage.h:
	Add support for multiple headers of the same type by storing the parsed
	headers in a GArray instaed of a hashtable.

1251 1252 1253 1254 1255 1256 1257
2007-05-21  Wim Taymans  <wim@fluendo.com>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
	(gst_udpsrc_create), (gst_udpsrc_unlock), (gst_udpsrc_unlock_stop):
	Since we depend on 0.10.13 -core, override the unlock_stop vmethod for
	safer shutdown.

1258 1259 1260 1261 1262 1263
2007-05-21  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_class_init):
	* gst/rtsp/gstrtpdec.h:
	Added signal for backwards compat.

1264 1265 1266 1267 1268 1269 1270 1271 1272 1273 1274 1275
2007-05-21  Sebastian Dröge  <slomo@circular-chaos.org>
	
	Patch by: René Stadler <mail at renestadler dot de>

	* configure.ac:
	* gst/auparse/gstauparse.c: (gst_au_parse_reset),
	(gst_au_parse_parse_header), (gst_au_parse_chain):
	* gst/auparse/gstauparse.h:
	Use audioconvert for converting from non-native endianness floats
	in auparse instead of doing it ourself. Fixes #424527.
	This needs the audioconvert from plugins-base CVS.
	
1276 1277 1278 1279 1280 1281
2007-05-21  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type),
	(gst_rtp_h263p_pay_flush):
	Fix enum registration.

1282 1283 1284 1285 1286 1287 1288 1289 1290 1291 1292
2007-05-21  Wim Taymans  <wim@fluendo.com>

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

	* gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type),
	(gst_rtp_h263p_pay_class_init), (gst_rtp_h263p_pay_init),
	(gst_rtp_h263p_pay_set_property), (gst_rtp_h263p_pay_get_property),
	(gst_rtp_h263p_pay_flush):
	* gst/rtp/gstrtph263ppay.h:
	Add new fragmentation mode base on GOB headers. Fixes #438940.

1293 1294 1295 1296 1297
2007-05-20  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp):
	  Printf format fix.

1298 1299 1300 1301 1302 1303
2007-05-18  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams):
	Don't crash when an unsupported transport error was returned by the
	server, just try to configure the next stream. Fixes #439255.

1304 1305 1306 1307 1308 1309 1310 1311 1312 1313 1314 1315 1316 1317 1318
2007-05-18  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
	(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
	(gst_rtspsrc_stream_configure_udp), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send),
	(gst_rtspsrc_setup_streams), (gst_rtspsrc_open):
	* gst/rtsp/gstrtspsrc.h:
	Add TCP timeout property and use it for all TCP connection.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
	(rtsp_connection_write), (rtsp_connection_next_timeout),
	(rtsp_connection_reset_timeout):
	Make connect and writes cancelable and make them use the timeout.

1319 1320 1321 1322 1323 1324 1325 1326 1327 1328 1329 1330 1331 1332 1333 1334
2007-05-18  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive),
	(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
	(gst_rtspsrc_try_send), (gst_rtspsrc_send),
	(gst_rtspsrc_setup_streams):
	Refactor timeout handling.
	Also send keep-alive when dealing with TCP transport.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	(rtsp_connection_free), (rtsp_connection_next_timeout),
	(rtsp_connection_reset_timeout):
	* gst/rtsp/rtspconnection.h:
	Use a timer to handle the session timeouts, add some methods to deal
	with timeouts.

1335 1336 1337 1338 1339 1340 1341 1342 1343 1344 1345
2007-05-17  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send),
	(gst_rtspsrc_setup_streams):
	Ignore streams that fail the setup command, we will retry with a
	different transport later on.

	* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp),
	(rtsp_ext_wms_configure_stream):
	Fix encoding name case.

1346 1347 1348 1349 1350
2007-05-16  Edward Hervey  <edward@fluendo.com>

	* ext/libpng/gstpngdec.c: (user_endrow_callback), (user_read_data):
	Fix build on macosx.

1351 1352 1353 1354 1355 1356
2007-05-16  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/raw1394/gstdv1394src.c: (gst_dv1394src_uri_set_uri):
	Replace direct comparison of a string with the string literal "" with
	a comparison of the first character with '\0'. Fixes #438926.

1357 1358 1359 1360 1361
2007-05-15  Stefan Kost  <ensonic@users.sf.net>

	* gst/debug/breakmydata.c (gst_break_my_data_init):
	  One more try. This should be the proper fix now.

1362 1363 1364 1365 1366
2007-05-15  Stefan Kost  <ensonic@users.sf.net>

	* gst/debug/breakmydata.c:
	  Ooops, no // comments please.

1367 1368 1369 1370 1371 1372
2007-05-15  Stefan Kost  <ensonic@users.sf.net>

	* gst/debug/breakmydata.c: (gst_break_my_data_class_init),
	(gst_break_my_data_init):
	  Fix gst_buffer_is_writable() assertion.

1373 1374 1375 1376 1377
2007-05-14  David Schleef  <ds@schleef.org>

	* sys/v4l2/gstv4l2src.c: Add support for Bayer images as
	  video/x-raw-bayer.  Fixes #314160.

1378 1379 1380 1381 1382 1383 1384 1385 1386 1387 1388 1389 1390
2007-05-14  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtptheoradepay.c: (decode_base64),
	(gst_rtp_theora_depay_parse_configuration):
	* gst/rtp/gstrtptheorapay.c: (encode_base64),
	(gst_rtp_theora_pay_finish_headers),
	(gst_rtp_theora_pay_handle_buffer):
	Update theora pay/depayloader in a similar to vorbis.

	* gst/rtp/gstrtpvorbisdepay.c:
	(gst_rtp_vorbis_depay_parse_configuration):
	Update docs.

1391 1392 1393 1394 1395 1396 1397
2007-05-14  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send):
	When we try to execute a method that is not supported by the server,
	don't error out but remove the method from the accepted methods so that
	we never try to perform this method again.

1398 1399 1400 1401 1402
2007-05-14  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_process):
	Remove annoying _dump_mem.

1403 1404 1405 1406 1407 1408 1409 1410
2007-05-14  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_range):
	Parse range correctly.

	* gst/rtsp/rtspurl.c: (rtsp_url_get_request_uri):
	The baseurl now always has a '/' at the start.

1411 1412 1413 1414 1415 1416 1417 1418 1419 1420 1421
2007-05-14  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_configure_caps),
	(gst_rtspsrc_parse_range), (gst_rtspsrc_open),
	(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
	Factor out caps configuration and configure more stuff such as the time
	ranges and speed/scale values.

	* gst/rtsp/rtsptransport.c:
	Add Copyright after non-trival fixes.

1422 1423 1424 1425 1426 1427 1428 1429 1430 1431 1432 1433 1434 1435 1436 1437 1438 1439 1440 1441 1442 1443
2007-05-12  Wim Taymans  <wim@fluendo.com>

	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/rtsp/gstrtspsrc.h:
	* gst/rtsp/rtspconnection.c: (rtsp_connection_receive):
	* gst/rtsp/rtspmessage.c: (rtsp_message_init_data),
	(rtsp_message_get_header):
	* gst/rtsp/rtspmessage.h:
	Make channel guint8 where possible.
	Make rtsp_message_init_data() take the channel as a guint8.

	* gst/rtsp/rtspdefs.c:
	Fixed a typo: Timout -> Timeout

	* gst/rtsp/rtspdefs.h:
	Make RTSP_CHECK() behave as a statement.

	* gst/rtsp/sdpmessage.c:
	Avoid a compiler warning in INIT_ARRAY().
	Fixes #437692.

1444 1445 1446 1447 1448 1449 1450 1451 1452
2007-05-12  Wim Taymans  <wim@fluendo.com>

	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_free),
	(rtsp_url_get_request_uri):
	* gst/rtsp/rtspurl.h:
	Add support for query parameters to RTSP URLs.

1453 1454 1455 1456 1457 1458 1459 1460 1461 1462 1463 1464 1465 1466 1467 1468 1469 1470
2007-05-12  Wim Taymans  <wim@fluendo.com>

	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
	(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
	(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
	(rtsp_transport_parse), (rtsp_transport_as_text):
	* gst/rtsp/rtsptransport.h:
	Add validation to rtsp_transport_parse().
	Add rtsp_transport_as_text() to generate an RTSP header from an
	RTSPTransport.
	Change ssrc to guint (was a string) since that is what it is, even
	though it is sent as a hex string.
	Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
	incorrect, which can be seen when looking at the examples in the RFC).
	Fixes #437670.

1471 1472 1473 1474 1475 1476 1477 1478 1479 1480
2007-05-11  Zaheer Abbas Merali  <<zaheerabbas at merali dot org>>

	Patch by: Eric Anholt

	* sys/ximage/gstximagesrc.c (gst_ximage_src_open_display,
	  gst_ximage_src_ximage_get):
	Use union of all damage between frames to make it faster.
	Fixes bug #342463.
	Also fix crasher when cursor is at bottom right of window.

1481 1482 1483 1484 1485 1486 1487
2007-05-11  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
	  Skip LIST chunks before the fmt chunk (fixes #437499). Also fix
	  streaming mode regression for file from #343837 with 'bext' chunk
	  before the 'fmt' chunk.

1488 1489 1490 1491 1492 1493 1494 1495 1496 1497 1498 1499 1500 1501 1502
2007-05-11  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
	(gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek),
	(gst_rtspsrc_handle_src_event),
	(gst_rtspsrc_stream_configure_manager),
	(gst_rtspsrc_stream_configure_tcp), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_open),
	(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
	* gst/rtsp/gstrtspsrc.h:
	* gst/rtsp/rtspdefs.h:
	Preliminary seek support.
	Activate internal pads so that we can receive events on them.
	Don't try to parse a range string when it's NULL.

1503 1504 1505 1506 1507 1508 1509 1510 1511 1512 1513 1514 1515 1516
2007-05-11  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/README:
	Update README with new RTP variables that will be used for
	synchronisation.

	* gst/rtp/gstrtpvorbisdepay.c: (decode_base64),
	(gst_rtp_vorbis_depay_parse_configuration),
	(gst_rtp_vorbis_depay_process):
	* gst/rtp/gstrtpvorbispay.c: (encode_base64),
	(gst_rtp_vorbis_pay_finish_headers),
	(gst_rtp_vorbis_pay_handle_buffer):
	Update vorbis pay and depayloader to draft-04.

1517 1518 1519 1520
2007-05-11  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/rtsptransport.c:
	UDP MCAST is actually the default for RTP/AVP.
1521 1522 1523 1524 1525
2007-05-13  Sebastien Moutte  <sebastien@moutte.net>

	* gst/level/gstlevel.c: (gst_level_transform_ip):
	Use guint8 * instead of gpointer then vs6 can build 
	in_data += (filter->width / 8).
1526

1527 1528 1529 1530 1531 1532 1533 1534 1535 1536
2007-05-11  Zaheer Abbas Merali  <<zaheerabbas at merali dot org>>

	* sys/ximage/gstximagesrc.c (gst_ximage_src_start,
	  gst_ximage_src_ximage_get):
	* sys/ximage/gstximagesrc.h (last_ximage):
	When using Damage actually keep the last frame, and not assume
	that the buffer we get already has the last frame on it.
	Copy the cursor over if we specify a non-zero start x and
	start y.

1537 1538 1539 1540 1541
2007-05-11  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/rtsptransport.c:
	Make UDP the default transport when not specified.

1542 1543 1544 1545 1546
2007-05-09  David Schleef  <ds@schleef.org>

	* gst/level/gstlevel.c:
	  Revert last change.

1547 1548 1549 1550 1551 1552 1553 1554 1555 1556 1557 1558 1559
2007-05-09  Sebastien Moutte  <sebastien@moutte.net>

	* gst/level/gstlevel.c: (gst_level_calculate_##TYPE),
	(gst_level_transform_ip):
	Use guint8 * instead of gpointer then vs6 know the size of data
	pointed when moving the pointer.
	* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_handle_buffer):
	Move instructions after variables declaration.
	* win32/vs6/autogen.dsp:
	* win32/vs6/libgstrtp.dsp:
	* win32/vs6/libgstrtsp.dsp:
	Update vs6 project files.

1560 1561 1562 1563 1564 1565 1566 1567 1568 1569 1570 1571
2007-05-09  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/Makefile.am:
	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_query),
	(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_open):
	* gst/rtsp/rtsprange.c: (parse_npt_time), (parse_npt_range),
	(parse_clock_range), (parse_smpte_range), (rtsp_range_parse),
	(rtsp_range_free):
	* gst/rtsp/rtsprange.h:
	Add code to parse time ranges.
	Report DURATION on the stream when possible.

1572 1573 1574 1575 1576 1577 1578
2007-05-08  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/videomixer/videomixer.c: (gst_videomixer_blend_ayuv_ayuv),
	(gst_videomixer_fill_checker), (gst_videomixer_fill_color),
	(gst_videomixer_collected):
	  Fix strides calculation for AYUV (it's just width*4) (#436910).

1579 1580 1581 1582 1583 1584 1585 1586
2007-05-06  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audiofx/audioamplify.c: (gst_audio_amplify_transform_ip):
	* gst/audiofx/audiodynamic.c: (gst_audio_dynamic_transform_ip):
	* gst/audiofx/audioinvert.c: (gst_audio_invert_transform_ip):
	Sync the GObject properties before each processing step to properly
	work with the controller.

1587 1588 1589 1590 1591 1592 1593 1594 1595 1596 1597 1598 1599
2007-05-04  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send),
	(gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
	(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
	(gst_rtspsrc_change_state):
	Let more error state trickle down so that we can catch more error
	cases.
	Handle keep-alive a little smarter by selecting a method the server
	actually supports.
	Fix a race in UDP streaming shutdown.

1600 1601 1602 1603 1604
2007-05-04  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive):
	Ignore errors when trying to use the keep-alive messages.

1605 1606 1607 1608 1609 1610 1611 1612 1613 1614 1615 1616 1617 1618 1619 1620 1621 1622 1623
2007-05-04  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_event),
	(gst_rtspsrc_handle_src_query), (gst_rtspsrc_sink_chain),
	(gst_rtspsrc_stream_configure_manager),
	(gst_rtspsrc_stream_free_udp), (gst_rtspsrc_stream_configure_tcp),
	(gst_rtspsrc_stream_configure_mcast),
	(gst_rtspsrc_stream_configure_udp),
	(gst_rtspsrc_stream_configure_udp_sink),
	(gst_rtspsrc_stream_configure_transport):
	Send RTCP messages back to the server over the TCP connection.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_write),
	(rtsp_connection_send), (rtsp_connection_read), (read_body),
	(rtsp_connection_receive):
	* gst/rtsp/rtspconnection.h:
	Factor out and expose lowlevel _write and _read methods.
	Implement sending data messages to the server.

1624 1625 1626 1627 1628 1629
2007-05-03  Wim Taymans  <wim@fluendo.com>

	* gst/multipart/multipartmux.c: (gst_multipart_mux_queue_pads),
	(gst_multipart_mux_collected):
	Fix timestamps on outgoing buffers.

1630 1631 1632 1633 1634 1635 1636
2007-05-03  Wim Taymans  <wim@fluendo.com>

	* gst/multipart/multipartmux.c:
	(gst_multipart_mux_request_new_pad), (gst_multipart_mux_collected),
	(gst_multipart_mux_change_state):
	Emit NEWSEGMENT events before pushing the first buffer.

1637 1638 1639 1640 1641 1642 1643 1644 1645 1646 1647 1648 1649 1650 1651 1652 1653 1654 1655 1656 1657
2007-05-03  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
	(gst_rtspsrc_alloc_udp_ports), (gst_rtspsrc_handle_src_event),
	(gst_rtspsrc_handle_src_query),
	(gst_rtspsrc_stream_configure_manager),
	(gst_rtspsrc_stream_free_udp), (gst_rtspsrc_stream_configure_tcp),
	(gst_rtspsrc_stream_configure_mcast),
	(gst_rtspsrc_stream_configure_udp),
	(gst_rtspsrc_stream_configure_udp_sink),
	(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_push_event),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_open),
	(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
	(gst_rtspsrc_pause):
	Refactor transport configuration code.
	Create internal pads for TCP transport so that we can implement events
	and queries.
	Handle events and queries.
	Parse range from the SDP.
	Fix race in pause handler where the connection could still be flushing.

1658 1659 1660 1661 1662 1663 1664 1665 1666 1667 1668 1669 1670 1671 1672 1673 1674 1675 1676
2007-05-02  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	(gst_rtspsrc_finalize), (new_session_pad), (request_pt_map),
	(gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
	(gst_rtspsrc_send), (gst_rtspsrc_async_open), (gst_rtspsrc_close),
	(gst_rtspsrc_play), (gst_rtspsrc_handle_message),
	(gst_rtspsrc_change_state):
	* gst/rtsp/gstrtspsrc.h:
	Fix race when multiple udp sources post timeouts, just act on the first
	received timeout.
	Protect stream list with a recursive lock to fix some races.
	Flush connection when we need to do a reconnect or stop.
	Make state lock recursive.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
	(rtsp_connection_close):
	Some small cleanups.

1677 1678 1679 1680 1681 1682 1683
2007-05-02  Wim Taymans  <wim@fluendo.com>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
	(gst_wavparse_stream_headers), (gst_wavparse_stream_data):
	Only set DISCONT when there actually is a discont or when we just
	started.

1684 1685 1686 1687 1688
2007-05-02  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/flac/gstflac.c: (plugin_init):
	Call bindtextdomain() to get localized strings.

1689 1690 1691 1692 1693 1694 1695 1696 1697 1698 1699
2007-05-02  Wim Taymans  <wim@fluendo.com>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
	(gst_wavparse_stream_headers), (gst_wavparse_add_src_pad),
	(gst_wavparse_stream_data):
	* gst/wavparse/gstwavparse.h:
	Be a bit more clever when dealing with VBR files with FACT tags, we
	don't want to timestamp buffers in that case but the estimated BPS can
	be used for seeking.
	Only send close segment in the streaming thread.

1700 1701 1702 1703 1704 1705 1706
2007-05-02  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/flac/gstflacdec.c: (gst_flac_dec_loop):
	Correctly post an error on the bus if something went wrong in the loop
	function. This fixes a few cases where the task was paused and nothing
	happened anymore.

1707 1708 1709 1710 1711 1712
2007-05-02  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/test.c: (main):
	Fix compilation of deprecated test just because I'm too lazy to delete
	it.

1713 1714 1715 1716 1717 1718 1719 1720 1721 1722 1723 1724 1725 1726 1727 1728 1729 1730 1731 1732 1733 1734 1735 1736 1737 1738 1739 1740 1741 1742 1743 1744
2007-05-02  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	(gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_handle_request), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_udp),
	(gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
	(gst_rtspsrc_open), (gst_rtspsrc_handle_message):
	* gst/rtsp/gstrtspsrc.h:
	Fix sending RTCP to the right place.
	Fix bug in reffing the wrong UDP element.
	Use new pad names for the session manager.
	Implement handling server requests in interleaved and UDP modes.
	Handle session keep-alive in UDP modes.
	Remove GCond for handling UDP timeouts.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
	(rtsp_connection_send), (rtsp_connection_read), (read_body),
	(rtsp_connection_receive), (rtsp_connection_close):
	* gst/rtsp/rtspconnection.h:
	Store connection IP address for later.
	Add timeout args to all operations that might block forever.
	Parse session timeout.
	Only close sockets when not already closed.

	* gst/rtsp/rtspdefs.c:
	* gst/rtsp/rtspdefs.h:
	Add timeout return value and error string.

	* gst/rtsp/rtspmessage.c: (rtsp_message_init_response):
	Add small comment.

1745 1746 1747 1748 1749 1750 1751 1752 1753
2007-05-01  Wim Taymans  <wim@fluendo.com>

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

	* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_init),
	(gst_rtp_mp4v_pay_empty), (gst_rtp_mp4v_pay_event):
	* gst/rtp/gstrtpmp4vpay.h:
	Handle NEWSEGMENT and FLUSH events. Fixes #434824.

1754 1755 1756 1757 1758 1759 1760 1761 1762
2007-04-30  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	  Remove v4l2src from docs, since it breaks the docs build, and the
	  plugin is only built if --enable-experimental is used anyway.

	* docs/plugins/Makefile.am:
	  Spaces => tab.

1763 1764 1765 1766 1767 1768 1769 1770 1771 1772 1773 1774 1775 1776 1777
2007-04-29  Wim Taymans  <wim@fluendo.com>

	* gst/udp/gstmultiudpsink.c: (leave_multicast),
	(gst_multiudpsink_add), (gst_multiudpsink_remove):
	Add code to drop membership of a multicast group.

	* gst/udp/gstudpsink.c: (gst_udpsink_update_uri),
	(gst_udpsink_set_uri):
	Implement URI handler.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_parse_rtpinfo):
	Use URI handler to make udpsink instace.
	Improve code to configure port and destination.

1778 1779 1780 1781 1782 1783 1784 1785 1786
2007-04-29  Wim Taymans  <wim@fluendo.com>

	* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_add):
	Fix multicast detection.
	Don't try to join a multicast group if the address is not multicast.

	* gst/udp/gstudpsrc.c: (gst_udpsrc_update_uri):
	Small debug improvement.

1787 1788 1789 1790 1791 1792 1793 1794
2007-04-27  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
	(gst_rtspsrc_handle_message):
	Ignore ASYNC state messages from the udpsink, it's irrelevant for the
	parent.

1795 1796 1797 1798 1799
2007-04-27  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpilbcdepay.h:
	Fix mode property when specified as an arg.

1800 1801 1802 1803 1804 1805 1806 1807
2007-04-26  Edward Hervey  <edward@fluendo.com>

	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	* docs/plugins/inspect/plugin-osxaudio.xml:
	Add documentation for osxaudio plugin.

1808 1809 1810 1811 1812 1813 1814 1815 1816 1817 1818 1819 1820 1821 1822
2007-04-26  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	(gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_open), (gst_rtspsrc_close),
	(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
	(gst_rtspsrc_pause):
	* gst/rtsp/gstrtspsrc.h:
	Protect state changes with a lock.

	* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
	(parse_line):
	* gst/rtsp/rtspconnection.h:
	Remove some unused stuff.

1823 1824 1825 1826 1827 1828
2007-04-26  Wim Taymans  <wim@fluendo.com>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
	Handle the case where there are exactly 0 bytes to read and the ioctl
	did not report an error. Fixes #433530.

1829 1830 1831 1832 1833 1834 1835 1836 1837
2007-04-26  Wim Taymans  <wim@fluendo.com>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
	(gst_wavparse_stream_headers), (gst_wavparse_stream_data):
	* gst/wavparse/gstwavparse.h:
	Apply DISCONT to buffers.
	Only apply timestamp to the first sample after a DISCONT, too many VBR
	files cause random jitter in the timestamps. Fixes #433119.

1838 1839 1840 1841 1842 1843 1844 1845 1846 1847 1848 1849 1850 1851 1852 1853 1854
2007-04-25  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_class_init),
	(gst_rtp_dec_init), (gst_rtp_dec_set_property),
	(gst_rtp_dec_get_property):
	* gst/rtsp/gstrtpdec.h:
	Add dummy latency property to be backwards compat with rtpbin.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
	(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
	(gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_parse_rtpinfo):
	* gst/rtsp/gstrtspsrc.h:
	Add latency property and configure in the session manager.
	Don't set invalid clock-base and seqnum-base on caps, some servers
	sometimes don't send them.

1855 1856 1857 1858 1859 1860 1861 1862 1863 1864 1865 1866 1867 1868 1869 1870 1871
2007-04-25  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/alpha/gstalphacolor.c: (gst_alpha_color_base_init),
	(gst_alpha_color_transform_caps), (gst_alpha_color_set_caps):
	  Double-check that RGB input caps are really RGBA caps (apparently
	  the core doesn't always catch it if those caps aren't a subset of
	  our template caps, also see #421543). Fixes #429319 in a way.
	  Also, don't leak the pad template in the transform_caps function.

	* tests/check/Makefile.am:
	* tests/check/elements/.cvsignore:
	* tests/check/elements/alphacolor.c: (setup_alphacolor),
	(cleanup_alphacolor), (create_caps_rgb24), (create_caps_rgba32),
	(create_buffer_rgb24_3x4), (create_buffer_rgba32_3x4),
	(GST_START_TEST), (alphacolor_suite):
	  Add some basic unit tests for alphacolor.

1872 1873 1874 1875 1876 1877 1878 1879
2007-04-25  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/libpng/gstpngdec.c: (gst_pngdec_task):
	  If we get a fatal flow return in the loop function, first post the
	  error message and only then send the EOS event downstream, otherwise
	  applications might get an eos message before the error message and
	  think everything was ok (related to #429319).

1880 1881 1882 1883 1884
2007-04-25  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/rtspconnection.c: (rtsp_connection_receive):
	Read the channel byte as an unsigned byte.

1885 1886 1887 1888 1889 1890 1891 1892 1893 1894 1895 1896 1897 1898 1899 1900 1901 1902 1903 1904 1905 1906
2007-04-25  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_set_property):
	* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_init),
	(gst_rtp_amr_depay_setcaps), (gst_rtp_amr_depay_process):
	* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_init),
	(gst_rtp_gsm_depay_setcaps):
	* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps):
	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps):
	* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_class_init),
	(gst_rtp_ilbc_depay_init), (gst_rtp_ilbc_depay_setcaps),
	(gst_rtp_ilbc_depay_process), (gst_ilbc_depay_set_property),
	(gst_ilbc_depay_get_property):
	* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps):
	* gst/rtp/gstrtpmp4adepay.c:
	* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_init),
	(gst_rtp_pcma_depay_setcaps):
	* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_init),
	(gst_rtp_pcmu_depay_setcaps):
	Make sure we configure the clock_rate in the baseclass in the setcaps
	function. Fixes #431282.

1907 1908 1909 1910 1911 1912 1913 1914 1915 1916 1917 1918 1919
2007-04-25  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
	(gst_rtspsrc_stream_free), (request_pt_map),
	(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open):
	* gst/rtsp/gstrtspsrc.h:
	Parse server address from SDP.
	Hook up a udpsink to send RTCP back to the server.

	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	* gst/rtsp/rtsptransport.h:
	Add some docs.

1920 1921 1922 1923 1924
2007-04-25  Stefan Kost  <ensonic@users.sf.net>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
	  Make header field check conditional. Fixes #433135

1925 1926 1927 1928 1929 1930 1931 1932 1933 1934 1935 1936
2007-04-24  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/inspect/plugin-alphacolor.xml:
	* gst/alpha/Makefile.am:
	* gst/alpha/gstalphacolor.c:
	* gst/alpha/gstalphacolor.h:
	  Add minimal docs blurb to alphacolor; split out headers into
	  separate header file for gtk-doc.

1937 1938 1939 1940 1941 1942
2007-04-20  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/debug/progressreport.c: (gst_progress_report_report):
	  Don't try to post NULL message (in case we can't query upstream
	  position or duration).

1943 1944 1945 1946 1947 1948 1949 1950
2007-04-18  Michael Smith  <msmith@fluendo.com>

	* gst/cutter/gstcutter.c: (gst_cutter_init), (gst_cutter_chain),
	(gst_cutter_get_caps):
	* gst/cutter/gstcutter.h:
	  Fix some of the most obvious bugs in cutter. Now doesn't leak
	  everything if input is silent.

1951 1952 1953 1954 1955 1956 1957 1958
2007-04-18  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
	(gst_wavenc_sink_setcaps), (gst_wavenc_change_state):
	* gst/wavenc/gstwavenc.h:
	Wav apparently only supports width==GST_ROUND_UP(depth), everything
	else results in a invalid block align and invalid files.

1959 1960 1961 1962 1963 1964 1965
2007-04-17  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Snaik <snaik32 gmail com>

	* gst/smpte/barboxwipes.c: (gst_wipe_boxes_draw):
	  Add missing break statement for BOX_HORIZONTAL case.

1966 1967 1968 1969 1970 1971 1972
2007-04-17  Sebastian Dröge  <slomo@circular-chaos.org>

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

	* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
	Use correct format strings for integer types.

1973 1974 1975 1976 1977 1978 1979 1980
2007-04-17  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
	(gst_wavparse_create_sourcepad):
	Use gst_riff_create_audio_template_caps () instead of the local caps.
	This makes updates of the local caps unecessary whenever libgstriff
	gets support for new formats.

1981 1982 1983 1984 1985 1986 1987 1988 1989 1990 1991 1992 1993 1994 1995 1996
2007-04-16  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Brian Cameron  <brian.cameron at sun dot com>

	* sys/sunaudio/gstsunaudio.c:
	* sys/sunaudio/gstsunaudiomixer.c:
	* sys/sunaudio/gstsunaudiomixer.h:
	* sys/sunaudio/gstsunaudiomixerctrl.c:
	* sys/sunaudio/gstsunaudiomixerctrl.h:
	* sys/sunaudio/gstsunaudiomixertrack.h:
	* sys/sunaudio/gstsunaudiosink.c:
	* sys/sunaudio/gstsunaudiosink.h:
	* sys/sunaudio/gstsunaudiosrc.c:
	* sys/sunaudio/gstsunaudiosrc.h:
	  Fix and/or update copyright attributions (#430228).

1997 1998 1999 2000 2001 2002 2003 2004 2005 2006 2007 2008 2009 2010 2011 2012 2013
2007-04-13  Wim Taymans  <wim@fluendo.com>

	* docs/plugins/gst-plugins-good-plugins-sections.txt:
	Fix docs.

	* gst/rtsp/URLS:
	Add some more example urls.

	* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT),
	(gst_rtp_dec_chain_rtp):
	Better debugging.

	* gst/rtsp/gstrtspsrc.c: (request_pt_map),
	(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_parse_rtpinfo):
	Remove unused code.

2014 2015 2016 2017 2018 2019 2020
2007-04-13  Stefan Kost  <ensonic@users.sf.net>

	* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
	(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
	(gst_wavparse_stream_data):
	  Relax the audio/mpeg caps again and add FIXME: comment.

2021 2022 2023 2024 2025 2026 2027 2028 2029
2007-04-13  Stefan Kost  <ensonic@users.sf.net>

	* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
	(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
	(gst_wavparse_stream_data):
	* gst/wavparse/gstwavparse.h:
	  More sanity check for the header fields. Fix type for 'rate' header
	  field.

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2007-04-12  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/icydemux/gsticydemux.c: (notgst_tag_freeform_string_to_utf8),
	(gst_icydemux_unicodify):
	  If the metadata strings we get in the stream are not UTF-8, try to
	  interpret them according to the character encodings specified in the
	  GST_ICY_TAG_ENCODING and GST_TAG_ENCODING environment variables, and
	  only fall back to locale/ISO-8859-1 if those aren't set or don't
	  work. Should fix #428901.

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2007-04-12  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtph264depay.c:
	Use the proper sync word for SPS and PPS.

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2007-04-12  Thomas Vander Stichele  <thomas at apestaart dot org>

	* gst/rtp/Makefile.am:
	* gst/rtp/fnv1hash.c (MASK_24, FNV1_HASH_32_INIT, FNV1_HASH_32_PRIME,
	  fnv1_hash_32_new, fnv1_hash_32_update, fnv1_hash_32_to_24):
	* gst/rtp/fnv1hash.h (__GST_FNV1_HASH_H__):
	  Add a simple hashing implementation that we can use to generate
	  a 24-bit ident value based on the codebooks for vorbis and theora.
	* gst/rtp/gstrtptheorapay.c (gst_rtp_theora_pay_finish_headers,
	  gst_rtp_theora_pay_handle_buffer):
	* gst/rtp/gstrtpvorbisdepay.c
	  (gst_rtp_vorbis_depay_parse_configuration,
	  gst_rtp_vorbis_depay_switch_codebook, gst_rtp_vorbis_depay_process):
	* gst/rtp/gstrtpvorbispay.c (gst_rtp_vorbis_pay_reset_packet,
	  gst_rtp_vorbis_pay_init_packet, gst_rtp_vorbis_pay_flush_packet,
	  gst_rtp_vorbis_pay_finish_headers, gst_rtp_vorbis_pay_handle_buffer):
	  Use the hashing function, ensuring that the same codebooks result
	  in the same ident and thus the same SDP description.
	  Various log fixes/changes.

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2007-04-12  Wim Taymans  <wim@fluendo.com>

	Patch by: jerry tan <jerry dot tan at sun dot com>

	* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_open):
	remove the call of  ioctl (fd, AUDIO_MIXER_MULTIPLE_OPEN), it is the
	application's responsibility to make sure it open the device once.
	Remove a careless error if AUDIODEV is set. Fixes #392620.

2074 2075 2076 2077 2078 2079 2080 2081 2082 2083 2084 2085 2086 2087 2088 2089
2007-04-12  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT),
	(gst_rtp_dec_class_init), (gst_rtp_dec_chain_rtp):
	* gst/rtsp/gstrtpdec.h:
	Make backward compat with rtpbin by adding the request-pt-map signals.

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
	(new_session_pad), (request_pt_map),
	(gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_stream_configure_caps),
	(gst_rtspsrc_activate_streams):
	* gst/rtsp/gstrtspsrc.h:
	Implement request-pt-map signals instead of setting caps on the buffers
	for the session manager.

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2007-04-11  Wim Taymans  <wim@fluendo.com>

	* gst/udp/gstudp.c: (plugin_init):
	Register GstNetBuffer in plugin_init so that the type can be used from
	multiple threads without races.

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2007-04-10  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps),
	(gst_rtp_amr_depay_process):
	Fix depayloader clock_rate and some cleanups.

	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_finalize),
	(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
	* gst/rtp/gstrtph264depay.h:
	Don't push codec_data in the adapter because it might get flushed when
	we get a discont.

	* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
	Handle multiple AU per packet.

	* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_process),
	(gst_rtp_sv3v_depay_plugin_init):
	Disable rank, this one does not work.
	Remove timestamping, base class does that.

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2007-04-10  Stefan Kost  <ensonic@users.sf.net>

	* gst/auparse/gstauparse.c: (gst_au_parse_parse_header):
	  limit caps to the formats we announce in the template

	* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
	(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
	(gst_wavparse_add_src_pad), (gst_wavparse_stream_data):
	  fix some crashers/asserts when dealing with broken files

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2007-04-10  Wim Taymans  <wim@fluendo.com>

	Patch by: Peter Kjellerstedt  <pkj at axis com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
	* gst/rtp/gstrtpL16depay.c:
	* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
	* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_get_mode),
	(gst_rtp_speex_depay_setcaps):
	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_loop_udp):
	* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init_send):
	Fix some compiler warnings. Fixes #428182.

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2007-04-06  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/Makefile.am:
	* gst/rtsp/gstrtpdec.c: (find_session_by_id), (create_session),
	(free_session), (gst_rtp_dec_base_init), (gst_rtp_dec_class_init),
	(gst_rtp_dec_init), (gst_rtp_dec_finalize),
	(gst_rtp_dec_query_src), (gst_rtp_dec_chain_rtp),
	(gst_rtp_dec_chain_rtcp), (gst_rtp_dec_set_property),
	(gst_rtp_dec_get_property), (gst_rtp_dec_provide_clock),
	(gst_rtp_dec_change_state), (create_recv_rtp), (create_recv_rtcp),
	(create_rtcp), (gst_rtp_dec_request_new_pad),
	(gst_rtp_dec_release_pad):
	* gst/rtsp/gstrtpdec.h:
	* gst/rtsp/gstrtsp.c: (plugin_init):
	Morph RTPDec into something compatible with RTPBin as a fallback.
	Various other style fixes.

	* gst/rtsp/gstrtspsrc.c: (find_stream_by_id),
	(find_stream_by_udpsrc), (gst_rtspsrc_stream_free),
	(gst_rtspsrc_cleanup), (gst_rtspsrc_media_to_caps),
	(new_session_pad), (gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_setup_auth),
	(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	* gst/rtsp/gstrtspsrc.h:
	Implement RTPBin session manager handling.
	Don't try to add empty properties to caps.
	Implement fallback session manager, handling.
	Don't combine errors from RTCP streams, just ignore them.

	* gst/rtsp/rtsptransport.c: (rtsp_transport_get_manager):
	* gst/rtsp/rtsptransport.h:
	Implement fallback session manager.
	Make RTPBin the default one when available.

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2007-04-05  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_class_init),
	(gst_rtp_mp4a_depay_setcaps), (gst_rtp_mp4a_depay_plugin_init):
	This element is ready to be autoplugged.

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2007-04-05  Julien MOUTTE  <julien@moutte.net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry):
	Don't leave the offsets defined by upstream element on the
	compressed data buffer we are pushing downstream. Make them
	GST_BUFFER_OFFSET_NONE.

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2007-04-04  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/README:
	* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
	(gst_avi_demux_parse_stream), (gst_avi_demux_parse_index),
	(gst_avi_demux_stream_index), (gst_avi_demux_sync),
	(gst_avi_demux_stream_scan), (gst_avi_demux_massage_index),
	(gst_avi_demux_calculate_durations_from_index),
	(gst_avi_demux_stream_header_push),
	(gst_avi_demux_stream_header_pull), (gst_avi_demux_combine_flows),
	(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data):
	  Don't abort on out-of-memory. Use stream-nr as unsigned integer only.

Wim Taymans's avatar
Wim Taymans committed
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2007-04-03  Wim Taymans  <wim@fluendo.com>

	* gst/smpte/barboxwipes.c:
	Fix error as spotted by Snaik <snaik32 at gmail dot com>

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2007-03-30  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/wavparse/gstwavparse.c:
	Support audio/x-raw-float in wav files. This only works with
	plugins-base CVS, using an older version doesn't have any
	disadvantages though.

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2007-03-30  Sebastian Dröge  <slomo@circular-chaos.org>

	* configure.ac:
	* gst/auparse/gstauparse.c: (gst_au_parse_reset),
	(gst_au_parse_parse_header), (gst_au_parse_chain):
	* gst/auparse/gstauparse.h:
	Revert last change as we don't want plugins-good to depend on
	plugins-base CVS now.

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2007-03-29  Sebastian Dröge  <slomo@circular-chaos.org>

	* configure.ac:
	Require gst-plugins-base CVS for audioconvert with non-native
	float support and width/depth fix in libgstriff.

	Patch by: René Stadler <mail at renestadler dot de>

	* gst/auparse/gstauparse.c: (gst_au_parse_reset),
	(gst_au_parse_parse_header), (gst_au_parse_chain):
	* gst/auparse/gstauparse.h:
	Don't swap the floats ourself if they're not in native endianness.
	Instead let audioconvert handle this. Fixes #339838.

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2007-03-29  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstasteriskh263.h:
	* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_process),
	(gst_rtp_h263p_depay_change_state):
	* gst/rtp/gstrtph263pdepay.h:
	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
	(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process),
	(gst_rtp_h264_depay_change_state):
	* gst/rtp/gstrtph264depay.h:
	* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_class_init),
	(gst_rtp_mp4a_depay_setcaps), (gst_rtp_mp4a_depay_process):
	* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
	Flush adapter on disconts.

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2007-03-29  Wim Taymans  <wim@fluendo.com>

	* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_process):
	* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_process):
	* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_process):
	* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_process):
	* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
	* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_flush):
	* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps),
	(gst_rtp_mp4v_depay_process):
	* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_flush):
	* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_process):
	* gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_flush):
	* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_process):
	* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_process):
	* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_process):
	* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_process):
	Use more efficient adapter and rtpbuffer methods when possible.

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2007-03-29  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
	(gst_wavenc_sink_setcaps):
	Correctly handle width!=depth input.
	* gst/wavparse/gstwavparse.c:
	Already export in the caps that width==8 uses unsigned samples and
	everything else uses signed samples.

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2007-03-29  Wim Taymans  <wim@fluendo.com>

	Patch by: Laurent Glayal <spglegle at yahoo dot fr>

	* gst/udp/gstdynudpsink.c: (gst_dynudpsink_class_init),
	(gst_dynudpsink_init), (gst_dynudpsink_set_property),
	(gst_dynudpsink_get_property), (gst_dynudpsink_init_send),
	(gst_dynudpsink_close):
	* gst/udp/gstdynudpsink.h:
	* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
	(gst_udpsrc_create), (gst_udpsrc_set_property),
	(gst_udpsrc_get_property), (gst_udpsrc_start), (gst_udpsrc_stop):
	* gst/udp/gstudpsrc.h:
	Rework the socket allocation a bit based on the sockfd argument so that
	it becomes usable.
	Add a closefd property to instruct the udp elements to close the custom
	file descriptors when going to READY. Fixes #423304.
	API:GstUDPSrc::closefd property
	API:GstDynUDPSink::closefd property

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2007-03-29  Wim Taymans  <wim@fluendo.com>

	Patch by: Laurent Glayal <spglegle at yahoo dot fr>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_base_init),
	(gst_rtp_h264_pay_class_init), (gst_rtp_h264_pay_init),
	(gst_rtp_h264_pay_finalize), (gst_rtp_h264_pay_setcaps),
	(gst_rtp_h264_pay_handle_buffer), (gst_rtp_h264_pay_set_property),
	(gst_rtp_h264_pay_get_property), (gst_rtp_h264_pay_change_state),
	(gst_rtp_h264_pay_plugin_init):
	* gst/rtp/gstrtph264pay.h:
	Added H264 payloader. Fixes #423782.

	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
	(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
	Small fixes.

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2007-03-29  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/wavparse/gstwavparse.c:
	Actually support depths from 1 to 32, not only 8 to 32.

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2007-03-29  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/wavparse/gstwavparse.c:
	Add support for wav files containing audio/x-raw-int with random
	depths between 1 and 32 bits.

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2007-03-28  Wim Taymans  <wim@fluendo.com>

	Based on patch by: Stefan Kost  <ensonic@users.sf.net>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_base_init),
	(gst_rtp_mp4a_depay_class_init), (gst_rtp_mp4a_depay_init),
	(gst_rtp_mp4a_depay_finalize), (gst_rtp_mp4a_depay_setcaps),
	(gst_rtp_mp4a_depay_process), (gst_rtp_mp4a_depay_set_property),
	(gst_rtp_mp4a_depay_get_property),
	(gst_rtp_mp4a_depay_change_state),
	(gst_rtp_mp4a_depay_plugin_init):
	* gst/rtp/gstrtpmp4adepay.h:
	Added MP4A-LATM depayloader. Fixes #417792.

	* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps),
	(gst_rtp_mp4v_depay_process):
	Fixup depayloader, setting codec_data, using more efficient adaptor and
	rtpbuffer handling.

	* gst/rtsp/URLS:
	Add url to test above.

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2007-03-25  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (find_stream_by_setup),
	(gst_rtspsrc_create_stream), (gst_rtspsrc_stream_free),
	(get_default_rate_for_pt), (gst_rtspsrc_parse_rtpmap),
	(gst_rtspsrc_media_to_caps),
	(gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_stream_configure_caps),
	(gst_rtspsrc_activate_streams), (gst_rtspsrc_parse_rtpinfo):
	* gst/rtsp/gstrtspsrc.h:
	Handle default clock-rates for static payload types, rearrange stuff so
	that the rtpmap field in the sdp can override the defaults.
	Parse RTP-Info field to get the seqnum and timebase fields that should
	go in the caps.
	Delay configuring caps after we got the RTP-Info from the PLAY reply from
	the server. 

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2007-03-22  Wim Taymans  <wim@fluendo.com>

	Patch by: Christophe Dehais <christophe dot dehais at gmail dot com>

	* ext/gconf/gconf.c: (gst_gconf_render_bin_with_default):
	Accept complex pipeline descriptions as an audio profile instead of just
	a single element. Fixes #420658.

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2007-03-21  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/apetag/gsttagdemux.c: (gst_tag_demux_get_type):
	  Rename registered type in preparation of GstTagDemux moving to
	  -base at some point in the future.

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2007-03-19  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
	  Streaming mode fixes: don't unref buffer we don't own any longer;
	  remove bogus adapter flush. Fixes #419338.

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2007-03-17  David Schleef  <ds@schleef.org>

	* REQUIREMENTS: Change the format to key/value, add a bunch of
	  information, remove a bunch of requirements that are for
	  other GStreamer packages.

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2007-03-17  David Schleef  <ds@schleef.org>

	* REQUIREMENTS: Fix a few things.  This file really needs a
	good once-over.

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2007-03-15  Edward Hervey  <edward@fluendo.com>

	* sys/Makefile.am:
	Don't forget to distribute the sys/osxaudio/ directory.

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2007-03-15  Edward Hervey  <edward@fluendo.com>

	* configure.ac:
	* sys/Makefile.am:
	* sys/osxaudio/Makefile.am:
	* sys/osxaudio/gstosxaudio.c:
	* sys/osxaudio/gstosxaudiosink.c:
	(gst_osx_audio_sink_osxelement_do_init), (gst_osx_audio_sink_init),
	(gst_osx_audio_sink_getcaps),
	(gst_osx_audio_sink_create_ringbuffer), (plugin_init):
	* sys/osxaudio/gstosxaudiosrc.c:
	(gst_osx_audio_src_osxelement_do_init), (gst_osx_audio_src_init),
	(gst_osx_audio_src_create_ringbuffer):
	* sys/osxaudio/gstosxringbuffer.c: (gst_osx_ring_buffer_get_type),
	(gst_osx_ring_buffer_class_init), (gst_osx_ring_buffer_init),
	(gst_osx_ring_buffer_acquire), (gst_osx_ring_buffer_start),
	(gst_osx_ring_buffer_pause), (gst_osx_ring_buffer_stop):
	* sys/osxaudio/gstosxringbuffer.h:
	Activate osxaudio in gst-plugins-good with proper build setup.
	Add inlined documentation.
	Fix debug statements
	Fix ringbuffer when pausing.
	Fixes #323471

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2007-03-14 Philippe Kalaf <philippe.kalaf@collabora.co.uk> 	 
	* gst/rtp/gstrtppcmapay.c:
	* gst/rtp/gstrtppcmapay.h:
	* gst/rtp/gstrtppcmupay.c:
	* gst/rtp/gstrtppcmupay.h:
	Ported mulaw and alaw payloaders to use new base class

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2007-03-14  Thomas Vander Stichele  <thomas at apestaart dot org>

	* po/af.po:
	* po/az.po:
	* po/cs.po:
	* po/en_GB.po:
	* po/it.po:
	* po/nl.po:
	* po/or.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/uk.po:
	* po/vi.po:
	  Update translations.

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2007-03-14  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  Fix string replace error (AG_AG_GST_* => AG_GST_*).

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2007-03-12  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/apetag/gsttagdemux.c: (gst_tag_demux_srcpad_event):
	  Fix handling of -1 values for start and stop values when seeking,
	  and SEEK_CUR+SEEK_END here as well.

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2007-03-12  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/id3demux/gstid3demux.c: (gst_id3demux_srcpad_event):
	  Fix handling of -1 values for start and stop values when seeking, 
	  and SEEK_CUR+SEEK_END.

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2007-03-12  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/id3demux/id3v2frames.c: (parse_picture_frame):
	  Fix parsing of ID3 v2.2.0 PIC frames. Only in version >= 2.3.0 is
	  the image format a variable-length NUL-terminated string; in
	  versions before that the image format is a fixed-length string of
	  3 characters (see #348644 for a sample tag).
	  Also make supplied mime type lower-case and fix up 'jpg' to 'jpeg'.

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2007-03-10  Sebastien Moutte  <sebastien@moutte.net>

	* win32/MANIFEST:
	Add new project files to MANIFEST.
	* win32/vs6/libgstaudiofx.dsp:
	* win32/vs6/libgstrtp.dsp:
	* win32/vs6/libgstrtsp.dsp:
	Update project files.
	
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2007-03-10  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/avi/gstavidemux.c: (gst_avi_demux_parse_subindex),
	(gst_avi_demux_parse_index):
	* sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_grab_frame):
	  Printf format fixes; also add some missing quotes in translated
	  strings. Fixes #416728 and #416727.

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2007-03-09  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_find_best):
	  Tim and I can't think of any reason the child audio sink needs to 
	  be set back to NULL after successfully determining that it can 
	  reach READY - it gets immediately set back to READY by the caller
	  anyway, causing an unnecessary close/open of any audio devices
	  involved.

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2007-03-09  Tim-Philipp Müller  <tim at centricular dot net>

	* po/LINGUAS:
	* po/ja.po:
	  Add ja.po file from #377306.

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2007-03-09  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/sunaudio/gstsunaudio.c: (plugin_init):
	* sys/sunaudio/gstsunaudiomixertrack.c:
	(gst_sunaudiomixer_track_new):
	  Actually translate sunaudio mixer track labels instead of just
	  marking the strings as translatable (#377306); clean up weird
	  label string mapping code that serves no apparent purpose. Also
	  set the 'untranslated-label' property when creating mixer tracks
	  if the GstMixerTrack base class supports this.

	* tests/check/Makefile.am:
	* tests/check/elements/.cvsignore:
	* tests/check/elements/sunaudio.c: (GST_START_TEST),
	(sunaudio_suite):
	  Very minimalistic unit test for sunaudiomixer element (compiles, but not
	  actually tested on a system where sunaudiomixer is available).

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2007-03-09  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/Makefile.am:
	Re-enable the states test and see if it works on the buildbots.

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2007-03-09  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_sink_setcaps),
	(gst_dvdec_src_negotiate), (gst_dvdec_chain),
	(gst_dvdec_change_state):
	* ext/dv/gstdvdec.h:
	Infer pixel-aspect-ratio from the video frame format if it isn't
	provided by the container, as happens when playing DV from AVI
	or Quicktime containers.

	Patch by: Wim Taymans <wim@fluendo.com>
	Fixes #380944

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2007-03-09  Wim Taymans  <wim@fluendo.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams):
	When activated, remove the udpsrc timeout, we have dataflow and timeouts
	will later be handled by the jitterbuffer.

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2007-03-09  Wim Taymans  <wim@fluendo.com>

	* ext/taglib/gstid3v2mux.cc:
	Add write support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
	Fixes #414496.
Jan Schmidt's avatar
Jan Schmidt committed
2559 2560
	
	Patch by: Alex Lancaster <alexl at users sourceforge net>
2561

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2007-03-09  Wim Taymans  <wim@fluendo.com>

	* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
	(gst_avi_demux_push_event), (gst_avi_demux_do_seek),
	(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
	(gst_avi_demux_chain):
	Fix stream position reporting after a seek. Fixes #416445.

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2007-03-08  Wim Taymans  <wim@fluendo.com>

	Patch by: René Stadler <mail at renestadler dot de>

	* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
	(gst_avi_demux_push_event), (gst_avi_demux_process_next_entry),
	(gst_avi_demux_stream_data), (gst_avi_demux_chain):
	Make avidemux accept optional header chunks in any order.
	Fixes #415446.

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2007-03-08  Jan Schmidt  <thaytan@mad.scientist.com>

	* tests/check/Makefile.am:
	Disable the states check until the remaining Valgrind errors
	are fixed or suppressed.

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2007-03-08  Sebastian Dröge  <slomo@circular-chaos.org>

	* tests/check/elements/.cvsignore:
	  Add audiodynamic check to .cvsignore

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2007-03-08  Sebastian Dröge  <slomo@circular-chaos.org>

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

	* gst/audiofx/Makefile.am:
	* gst/audiofx/audiodynamic.c:
	(gst_audio_dynamic_characteristics_get_type),
	(gst_audio_dynamic_mode_get_type),
	(gst_audio_dynamic_set_process_function),
	(gst_audio_dynamic_base_init), (gst_audio_dynamic_class_init),
	(gst_audio_dynamic_init), (gst_audio_dynamic_set_property),
	(gst_audio_dynamic_get_property), (gst_audio_dynamic_setup),
	(gst_audio_dynamic_transform_hard_knee_compressor_int),
	(gst_audio_dynamic_transform_hard_knee_compressor_float),
	(gst_audio_dynamic_transform_soft_knee_compressor_int),
	(gst_audio_dynamic_transform_soft_knee_compressor_float),
	(gst_audio_dynamic_transform_hard_knee_expander_int),
	(gst_audio_dynamic_transform_hard_knee_expander_float),
	(gst_audio_dynamic_transform_soft_knee_expander_int),
	(gst_audio_dynamic_transform_soft_knee_expander_float),
	(gst_audio_dynamic_transform_ip):
	* gst/audiofx/audiodynamic.h:
	* gst/audiofx/audiofx.c: (plugin_init):
	Add new audiodynamic element which can act as a compressor or
	expander. Supported are hard-knee and soft-knee operation modes with
	user-specified ratio and threshold.
	Attack and release parameters are not yet implemented but will follow.
	* 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:
	Integrate audiodynamic into the docs.
	* tests/check/Makefile.am:
	* tests/check/elements/audiodynamic.c: (setup_dynamic),
	(cleanup_dynamic), (GST_START_TEST), (dynamic_suite), (main):
	Add unit test for audiodynamic.

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2007-03-07  Jan Schmidt  <thaytan@mad.scientist.com>

	* ext/raw1394/gstdv1394src.c: (gst_dv1394src_start):
	Free handles that we allocated when exiting via the error paths.

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2007-03-07  Stefan Kost  <ensonic@users.sf.net>

	* gst/level/gstlevel.c: (gst_level_class_init),
	(gst_level_set_caps), (gst_level_start), (gst_level_event),
	(gst_level_transform_ip):
	* gst/level/gstlevel.h:
	  Resolve message timestamps against the playback segment.