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2007-12-18  Tim-Philipp Müller  <tim at centricular dot net>

	* Makefile.am:
	  Include common/win32.mak for CRLF check of win32 project
	  files (see #393626).

	* win32/vs6/libgstpng.dsp:
	  Fix line endings and do cvs admin -kb.

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2007-12-17  Edward Hervey  <edward.hervey@collabora.co.uk>

	* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_chain):
	Actually drop the buffers which are outside the currently configured
	segment instead of just emitting a WARNING.

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2007-12-14  Wim Taymans  <wim.taymans@collabora.co.uk>

	* ext/flac/gstflacdec.c: (gst_flac_dec_metadata_callback),
	(gst_flac_dec_write):
	* ext/flac/gstflacdec.h:
	Send segments from the streaming thread. Fixes #502187.
	Fix segment seeking and a bunch of other seeking cases.

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2007-12-14  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/id3demux/id3v2frames.c: (parse_url_link_frame):
	  Parse WOAF frames and put the result into GST_TAG_CONTACT,
	  which is where it would end up if the same information was
	  put in a vorbis comment (don't think it's worth adding a
	  new URI tag for this). Fixes #488112.

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2007-12-11  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  We need core/base 0.10.15 or later.

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2007-12-11  Wim Taymans  <wim.taymans@collabora.co.uk>

	Patch by: Mark Nauwelaerts <manauw skynet be>

	* gst/avi/gstavimux.c: (gst_avi_mux_start_file):
	Fix regression in stream numbering. Fixes #502655.

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2007-12-11  Wim Taymans  <wim.taymans@collabora.co.uk>

	Patch by: Wai-Ming Ho <webregbox at yahoo dot co dot uk>

	* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_init),
	(gst_rtp_h264_pay_finalize), (gst_rtp_h264_pay_setcaps),
	(next_start_code), (is_nal_equal), (gst_rtp_h264_pay_decode_nal),
	(encode_base64), (gst_rtp_h264_pay_parse_sps_pps),
	(gst_rtp_h264_pay_handle_buffer):
	* gst/rtp/gstrtph264pay.h:
	Use higher performance start-code searching.
	Parse NALs and store SPS, PPS and profile in the caps so that they can
	be used in the SDP. Fixes #502814.

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2007-12-11  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list):
	  Init some structs to zero before we pass them to ioctl, which
	  avoids valgrind warnings.  Also fix a small memory leak.

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2007-12-11  Wim Taymans  <wim.taymans@collabora.co.uk>

	Patch by: Wouter Cloetens <wouter at mind dot be>

	* gst/multipart/multipartdemux.c: (gst_multipart_demux_chain):
	Copy timestamp from input to output. Not very perfect yet but better
	than nothing. Fixes #503023.

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2007-12-09  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
	  Also print a useful error message with the old Wavpack API
	  if possible.

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2007-12-09  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/wavpack/gstwavpackdec.c:
	  More build fixes for old libwavpack versions: include config.h so
	  that WAVPACK_OLD_API is actually defined as detected; only use
	  WavpackGetErrorMessage if it is available. This fixes the build
	  on debian stable for me.

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2007-12-09  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
	* ext/wavpack/gstwavpackparse.c:
	  (gst_wavpack_parse_create_src_pad):
	  Workaround the non-existance of WavpackGetChannelMask in Wavpack
	  versions below 4.40.0.

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2007-12-09  Sebastian Dröge  <slomo@circular-chaos.org>

	* configure.ac:
	  And now do it right for real...

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2007-12-09  Sebastian Dröge  <slomo@circular-chaos.org>

	* configure.ac:
	  Correctly reset $LIBS to not contain -lm.

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2007-12-09  Sebastian Dröge  <slomo@circular-chaos.org>

	Based on a patch by: Kwang Yul Seo <kwangyul dot seo at gmail dot com>

	* configure.ac:
	* ext/cairo/gsttimeoverlay.c:
	  (gst_cairo_time_overlay_print_smpte_time):
	  Fix compilation with MSVC by using gst_util_guint64_to_gdouble()
	  and checking for rint() and implementing it ourself if it doesn't
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	  exist. Fixes #497293.
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2007-12-09  Sebastian Dröge  <slomo@circular-chaos.org>

	* configure.ac:
	  Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.

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2007-12-08  Jan Schmidt  <jan.schmidt@sun.com>

	* sys/oss/gstosshelper.c:
	Verify that the format returned after the ioctl is the one
	we requested. It is valid for the ioctl to succeed while
	substituting an alternate 'supported' sample format.

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2007-12-07  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/oss/gstossaudio.c: (plugin_init):
	* sys/oss/gstosssink.c: (gst_oss_sink_open):
	* sys/oss/gstosssrc.c: (gst_oss_src_open):
	  Post decent (and translated) error message when we can't
	  open the audio device for some reason.

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2007-12-07  Jan Schmidt  <jan.schmidt@sun.com>

	* sys/oss/gstosssink.c:
	* sys/oss/gstosssrc.c:
	Allow the AUDIODEV environment variable to redirect us
	to a different default OSS device, like sunaudiosink does
	on Solaris (makes audio play automatically on SunRays).

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2007-12-06  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audiofx/audioamplify.c: (gst_audio_amplify_transform_ip):
	* gst/audiofx/audiochebyshevfreqband.c:
	(gst_audio_chebyshev_freq_band_transform_ip):
	* gst/audiofx/audiochebyshevfreqlimit.c:
	(gst_audio_chebyshev_freq_limit_transform_ip):
	* gst/audiofx/audiodynamic.c: (gst_audio_dynamic_transform_ip):
	* gst/audiofx/audioinvert.c: (gst_audio_invert_transform_ip):
	The transform_ip() methods should do nothing if in passthrough mode.
	It might get non-writable buffers in that case but the buffer might
	as well be writable.

	* gst/audiofx/audiopanorama.c: (gst_audio_panorama_transform):
	The transform() methods won't be called in passthrough mode and
	otherwise the buffer is always writable so don't check here.

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2007-12-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_srcpad_event):
	  Fix seeking in .wav files again (#501775).  Some people seem to think
	  they don't need to test their changes when they're just 'reflowing'
	  some code.

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2007-12-05  Wim Taymans  <wim.taymans@gmail.com>

	* gst/autodetect/gstautovideosink.c:
	(gst_auto_video_sink_class_init), (gst_auto_video_sink_dispose),
	(gst_auto_video_sink_init),
	(gst_auto_video_sink_create_element_with_pretty_name),
	(gst_auto_video_sink_find_best),
	(gst_auto_video_sink_set_property),
	(gst_auto_video_sink_get_property):
	* gst/autodetect/gstautovideosink.h:
	Fix docs.
	Use same error reporting code as autoaudiosink.
	Add property to filter sinks based on caps. Only select raw video sinks
	by default for backwards compat.
	API: GstAutoVideoSink::filter-caps

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2007-12-05  Wim Taymans  <wim.taymans@gmail.com>

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

	* gst/autodetect/gstautoaudiosink.c:
	(gst_auto_audio_sink_class_init), (gst_auto_audio_sink_dispose),
	(gst_auto_audio_sink_init), (gst_auto_audio_sink_find_best),
	(gst_auto_audio_sink_set_property),
	(gst_auto_audio_sink_get_property):
	* gst/autodetect/gstautoaudiosink.h:
	Add property to filter sinks based on caps. Only select raw audio sinks
	by default for backwards compat.  Fixes #417420.
	API: GstAutoAudioSink::filter-caps

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2007-11-29  Michael Smith <msmith@fluendo.com>

	Patch by: Arek Korbik <arkadini@gmail.com>

	* gst/videobox/gstvideobox.c: (plugin_init):
	  Initialise liboil in plugin_init()

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2007-11-29  Wim Taymans  <wim.taymans@gmail.com>

	* ext/libpng/gstpngdec.c: (gst_pngdec_task):
	Post error before sending EOS. Fixes #499178.

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2007-11-28  Sebastien Moutte  <sebastien@moutte.net>

	* win32/vs6/gst_plugins_good.dsw:
	* win32/vs6/libgstpng.dsp:
	Add a project file for libgstpng

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2007-11-28  Edward Hervey  <bilboed@bilboed.com>

	* gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_class_init),
	(gst_rtp_h263_depay_process):
	Code beautification.
	Added debug statements.
	Don't bit-shift everything, just do operations on last/first byte
	instead.

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2007-11-27  Wim Taymans  <wim.taymans@gmail.com>

	Patch by: Jayarama S. Santana <sundarsantana at gmail dot com>

	* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_process):
	Fix wrong comparison in overrun check. Fixes #499239 some more.

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2007-11-27  Edward Hervey  <bilboed@bilboed.com>

	* gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_init),
	(gst_rtp_h263_depay_process):
	* gst/rtp/gstrtph263depay.h:
	Fix h263 depayloader so that ANY h263 decoder can handle the outgoing
	stream.

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2007-11-26  Wim Taymans  <wim.taymans@gmail.com>

	Based on Path by: Jayarama S. Santana <sundarsantana at gmail dot com>

	* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_setcaps),
	(gst_rtp_mp4a_depay_process):
	* gst/rtp/gstrtpmp4adepay.h:
	Fix depayloading when multiple frames are inside one RTP packet.
	Fixes #499239.

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2007-11-26  Stefan Kost  <ensonic@users.sf.net>

	* gst/level/gstlevel.c:
	  Add GAP-flag support.

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2007-11-26  Edward Hervey  <bilboed@bilboed.com>

	* gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_process):
	Read the I flag for Mode A h263 rtp stream and set the
	GST_BUFFER_FLAG_DELTA_UNIT accordingly.
	Fixes #499383

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2007-11-26  Stefan Kost  <ensonic@users.sf.net>

	* gst/level/gstlevel.c:
	  Remove some dead code and do cleanups.

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2007-11-26  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/pipelines/simple-launch-lines.c:
	  Improve the tests by allowing to set a target state.

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2007-11-26  Sebastian Dröge  <slomo@circular-chaos.org>

	* tests/check/elements/wavpackenc.c: (GST_START_TEST):
	Don't check the caps of the output buffer if they're equal some
	other caps. The caps can change in a backward compatible way
	and did at this point.

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2007-11-24  Julien MOUTTE  <julien@moutte.net>

	* gst/qtdemux/qtdemux.c: (gst_qtdemux_find_segment),
	(gst_qtdemux_move_stream), (gst_qtdemux_do_seek),
	(gst_qtdemux_seek_to_previous_keyframe),
	(gst_qtdemux_activate_segment), (gst_qtdemux_advance_sample),
	(gst_qtdemux_loop_state_movie), (gst_qtdemux_loop): Implement
	reverse playback support.

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2007-11-20  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/wavpack/gstwavpackcommon.c: (gst_wavpack_set_channel_layout):
	Also set the channel layout on the Wavpack caps if we're having
	a mono layout. Of course only do it for "audio/x-wavpack".

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2007-11-20  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/wavpack/gstwavpackcommon.c:
	(gst_wavpack_get_default_channel_mask),
	(gst_wavpack_set_channel_layout),
	(gst_wavpack_get_default_channel_positions),
	(gst_wavpack_get_channel_mask_from_positions),
	(gst_wavpack_set_channel_mapping):
	* ext/wavpack/gstwavpackcommon.h:
	* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
	(gst_wavpack_dec_sink_set_caps), (gst_wavpack_dec_chain):
	* ext/wavpack/gstwavpackdec.h:
	* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_reset),
	(gst_wavpack_enc_init), (gst_wavpack_enc_sink_set_caps),
	(gst_wavpack_enc_set_wp_config), (gst_wavpack_enc_push_block),
	(gst_wavpack_enc_fix_channel_order), (gst_wavpack_enc_chain),
	(gst_wavpack_enc_rewrite_first_block),
	(gst_wavpack_enc_sink_event):
	* ext/wavpack/gstwavpackenc.h:
	* ext/wavpack/gstwavpackparse.c:
	(gst_wavpack_parse_index_append_entry), (gst_wavpack_parse_reset),
	(gst_wavpack_parse_scan_to_find_sample),
	(gst_wavpack_parse_sink_event), (gst_wavpack_parse_create_src_pad),
	(gst_wavpack_parse_push_buffer), (gst_wavpack_parse_loop):
	* ext/wavpack/gstwavpackparse.h:
	Add support for encoding, parsing and decoding multichannel
	files with up to 8 channels. This also improves the robustness
	of parsing quite a bit.

	* ext/wavpack/gstwavpackstreamreader.c:
	(gst_wavpack_stream_reader_read_bytes),
	(gst_wavpack_stream_reader_get_pos),
	(gst_wavpack_stream_reader_set_pos_abs),
	(gst_wavpack_stream_reader_set_pos_rel),
	(gst_wavpack_stream_reader_push_back_byte),
	(gst_wavpack_stream_reader_get_length),
	(gst_wavpack_stream_reader_can_seek),
	(gst_wavpack_stream_reader_write_bytes):
	Improve debugging.

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2007-11-20  Stefan Kost  <ensonic@users.sf.net>

	* ext/libpng/gstpngdec.c:
	* ext/libpng/gstpngdec.h:
	  Don't release the png-memory from within the callback.

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2007-11-20  Stefan Kost  <ensonic@users.sf.net>

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

	* ext/libpng/gstpngenc.c:
	  Don't leak buffer data memory. Fixes #498395.

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2007-11-20  Stefan Kost  <ensonic@users.sf.net>

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

	* tests/check/pipelines/simple-launch-lines.c:
	  Tests for #498395.

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2007-11-20  Julien MOUTTE  <julien@moutte.net>

	* ext/taglib/gsttaglibmux.c: (gst_tag_lib_mux_render_tag),
	(gst_tag_lib_mux_adjust_event_offsets):
	* gst/qtdemux/qtdemux.c: (qtdemux_parse_theora_extension):
	* sys/osxaudio/Makefile.am:
	* sys/osxvideo/cocoawindow.h:
	* sys/osxvideo/cocoawindow.m: Fix build on Mac OS X 10.5

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2007-11-15  David Schleef  <ds@schleef.org>

	* ext/cairo/gsttextoverlay.c:
	  Change strcasecmp() to g_strcasecmp().  Fixes #497292.

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2007-11-15  Wim Taymans  <wim.taymans@gmail.com>

	Patch by: Jordi Jaen Pallares <jordijp at gmail dot com>

	* gst/rtp/gstrtpmp2tpay.c: (gst_rtp_mp2t_pay_class_init),
	(gst_rtp_mp2t_pay_init), (gst_rtp_mp2t_pay_finalize),
	(gst_rtp_mp2t_pay_flush), (gst_rtp_mp2t_pay_handle_buffer):
	* gst/rtp/gstrtpmp2tpay.h:
	Fill the MTU with as many packets as possible. Fixes #491323.

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2007-11-15  Wim Taymans  <wim.taymans@gmail.com>

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

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams):
	Fix some more leaks. Fixes #497007.

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2007-11-15  Wim Taymans  <wim.taymans@gmail.com>

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

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_free),
	(gst_rtspsrc_stream_configure_tcp):
	Fix 3 pad leaks. Fixes #496983.

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2007-11-15  Wim Taymans  <wim.taymans@gmail.com>

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

	* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps):
	Fix small leak. Fixes #497017.

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2007-11-15  Wim Taymans  <wim.taymans@gmail.com>

	* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
	(gst_qtdemux_prepare_current_sample),
	(gst_qtdemux_loop_state_movie), (qtdemux_parse_theora_extension),
	(qtdemux_parse_node), (qtdemux_parse_trak), (qtdemux_video_caps):
	* gst/qtdemux/qtdemux_fourcc.h:
	* gst/qtdemux/qtdemux_types.c:
	Add suppport for theora in quicktime according to XiphQT.

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2007-11-15  Edgard Lima  <edgard.lima@indt.org.br>

	* sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init),
	(gst_v4l2src_init), (gst_v4l2src_set_property),
	(gst_v4l2src_get_property):
	* sys/v4l2/gstv4l2src.h:
	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_grab_frame):
	  Always copy buffers by default (handle safer with bugged drivers)
	  and added a property to make it possible to use mmap effectively (no
	  copy if possible) when application wants to. Fixes: #480557.

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2007-11-14  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/id3demux/id3tags.c:
	* gst/id3demux/id3tags.h:
	* gst/id3demux/id3v2frames.c: (id3v2_tag_to_taglist):
	  We don't want the same string multiple times in a tag list for the
	  same tag ever, for any tag, not just for GST_TAG_GENRE, so make sure
	  this doesn't happen and remove special-case code for GST_TAG_GENRE.

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2007-11-14  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/taglib/gstid3v2mux.cc: (add_musicbrainz_tag), (add_funcs):
	  Write GST_TAG_MUSICBRAINZ_DISCID and GST_TAG_CDDA_CDDB_DISCID
	  into ID3v2 TXXX frames (fixes #347848).

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2007-11-14  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open):
	  Don't leak sdp message contents (fixes #496773).

	* gst/udp/gstudpsink.c: (gst_udpsink_finalize):
	  Don't leak URI string.

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2007-11-14  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Julien Puydt <julien dot puydt at laposte net>

	* ext/raw1394/Makefile.am:
	* ext/raw1394/gst1394probe.c: (gst_1394_get_guid_array),
	  (gst_1394_property_probe_get_properties),
	  (gst_1394_property_probe_probe_property),
	  (gst_1394_property_probe_needs_probe),
	  (gst_1394_property_probe_get_values),
	  (gst_1394_property_probe_interface_init),
	  (gst_1394_type_add_property_probe_interface):
	* ext/raw1394/gst1394probe.h: (GST_1394_PROBE_H):
	* ext/raw1394/gstdv1394src.c: (_do_init), (gst_dv1394src_class_init),
	  (gst_dv1394src_init), (gst_dv1394src_dispose),
	  (gst_dv1394src_set_property), (gst_dv1394src_get_property),
	  (gst_dv1394src_discover_avc_node), (gst_dv1394src_query),
	  (gst_dv1394src_update_device_name):
	* ext/raw1394/gstdv1394src.h:
	  Implement GstPropertyProbe interface and add "device-name" property,
	  so applications can use this to probe for available devices in the
	  same way they can already with v4lsrc and v4l2src (however horrible
	  this property probe interface may be). Fixes #358841.

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2007-11-14  Tim-Philipp Müller  <tim at centricular dot net>

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

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_event),
	  (gst_rtspsrc_parse_range):
	  Don't leak event, don't leak range (fixes #496752).

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2007-11-14  Michael Smith <msmith@fluendo.com>

	Patch by: Arek Korbik <arkadini@gmail.com>

	* gst/alpha/gstalphacolor.c: (gst_alpha_color_set_caps):
	  Detect RGBA/BGRA correctly on little endian systems.

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2007-11-13  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_probe_caps_for_format):
	  If VIDIOC_ENUM_FRAMESIZES is defined (= recent kernel), but the
	  corresponding ioctl() call fails even though the driver claims to
	  support this format, just fall back to the pre-2.6.19 kernel
	  routine that creates caps with suitable height and width ranges
	  (see #448278).

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2007-11-13  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Mark Nauwelaerts <manauw skynet be>

	* gst/matroska/matroska-demux.c:
	  (gst_matroska_demux_push_dvd_clut_change_event),
	  (gst_matroska_demux_parse_blockgroup_or_simpleblock),
	  (gst_matroska_demux_subtitle_caps):
	* gst/matroska/matroska-ids.h:
	  Extract palette data for dvd subpicture streams and send it
	  downstream as custom gstreamer dvd event (fixes #453417).

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2007-11-13  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/cairo/gsttextoverlay.c: (gst_text_overlay_font_init):
	  Implement minimal parsing of the passed pango font description
	  string, so passing a font size works the same as with the
	  pango textoverlay plugin; fixes #455086.
	  (Maybe we could just use pangocairo here at some point).

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2007-11-13  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c:
	* gst/wavparse/gstwavparse.c:
	  Return the result in _activate_pull(). Don't ref element there.

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2007-11-13  Stefan Kost  <ensonic@users.sf.net>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers),
	(gst_wavparse_pad_convert), (gst_wavparse_pad_query),
	(gst_wavparse_srcpad_event):
	  Ref the element when we should, but not when we its not needed. Reflow
	  the event_handling to not leak the event.	  

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2007-11-12  Edward Hervey  <bilboed@bilboed.com>

	* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
	(qtdemux_parse_samples):
	Properly free QTDemuxSamples array.
	Protect table write with a sensible check, some files apparently DO contain
	stts values starting with 0 :(

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2007-11-12  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c:
	* gst/qtdemux/qtdemux.c:
538
	  Drop QOS in _handle_src_event(). Fix the refcount in qtdemux that
539 540
	  previous commit messed up.

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2007-11-12  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c:
	* gst/qtdemux/qtdemux.c:
	  Sync _handle_src_event() with oggdemux. In avidemux also ref the
	  element when we should, but not when we its not needed.

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2007-11-08  Stefan Kost  <ensonic@users.sf.net>

	* gst/wavparse/gstwavparse.c:
	  Return FALSE if we can't handle a query instead of changing the
	  format. Ignore fact when dealing with mpeg audio.

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2007-11-02  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Ole André Vadla Ravnås  <ole.andre.ravnas@tandberg.com>

	* configure.ac:
	* gst/udp/gstdynudpsink.c:
	* gst/udp/gstdynudpsink.h:
	* gst/udp/gstmultiudpsink.c:
	* gst/udp/gstmultiudpsink.h:
	* gst/udp/gstudpsink.c:
	* gst/udp/gstudpsink.h:
	  Fix includes for MSVC and GLib-2.14.0 (#492388).

	* gst/udp/gstudpsrc.c: (gst_udpsrc_start):
	  No more pipe define since GLib-2.14.0, need to use _pipe() directly.

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2007-11-02  Edward Hervey  <bilboed@bilboed.com>

	* gst/law/mulaw-decode.c: (mulawdec_sink_setcaps),
	(gst_mulawdec_chain):
	* gst/law/mulaw-decode.h:
	Calculate outgoing buffer duration if incoming buffer didn't have a
	valid duration.

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2007-10-30  Wim Taymans  <wim.taymans@gmail.com>

	* gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment),
	(gst_qtdemux_combine_flows), (gst_qtdemux_loop_state_movie):
	Smarter combine_flow code that also deals with downstream elements
	returning UNEXPECTED when they receive data out of the segment
	boundaries. Fixes #491305.

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2007-10-26  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/v4l2/v4l2src_calls.c:
	  Fix 'unused variable' compiler warning when compiling against
	  older kernel headers.

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2007-10-25  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/taglib/gstid3v2mux.cc (add_funcs):
	  Map new SORTNAME tags to ID3v2 TSOP, TSOA and TSOT frames (#414539).

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2007-10-24  Stefan Kost  <ensonic@users.sf.net>

	* tests/check/pipelines/simple-launch-lines.c:
	   Improve the tests a little more.

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2007-10-23  Zaheer Abbas Merali <zaheerabbas at merali dot org>

	patch by: Yun Zheng Hu

	* sys/osxaudio/gstosxaudiosrc.c:
	Use default input device instead of default output device and
	only memcpy actual available bytes.

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2007-10-22  Edgard Lima  <edgard.lima@indt.org.br>

	* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_grab_frame):
	  Fixes "v4l2src ! queue ! xvimagesink". The queue ask for buffer too
	  early. It is temporary until we find something better.

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2007-10-22  Wim Taymans  <wim.taymans@gmail.com>

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

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved):
	Fix race when pausing a RTSP stream in interleaved.
	Fixes #475784.

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2007-10-22  Wim Taymans  <wim.taymans@gmail.com>

	Patch by: Peter Kjellerstedt <pkj at axis com>

	* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_finalize):
	Use correct unref function for buffers. #488844.

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2007-10-19  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavimux.c:
	* tests/check/elements/avimux.c:
	  Add some debug and sync tests with the fix.	  

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2007-10-18  Wim Taymans  <wim.taymans@gmail.com>

	Based on patch by: Laurent Glayal  <spglegle yahoo fr>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
	When the socket is used by the app for other purposes, don't generate an
	error if there is activaty on the socket that is not data related.
	Fixes #487488.

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2007-10-18  Wim Taymans  <wim.taymans@gmail.com>

	* sys/v4l2/v4l2src_calls.c: (gst_v4l2_buffer_finalize),
	(gst_v4l2src_grab_frame):
	Add some more debug info. Generate an error when we run out of buffers
	for some reason. See #480557.

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2007-10-18  Wim Taymans  <wim.taymans@gmail.com>

	Patch by: Anders Skargren <anders dot skargren at axis dot com>

	* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_handle_buffer):
	Set marker bit correctly.

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2007-10-17  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open):
	Use allowed name for the GstStructure.

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2007-10-17  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/gconf/gstswitchsink.c:
	* gst/autodetect/gstautoaudiosink.c:
	  Use new gst_bus_pop_filtered().

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2007-10-13  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/v4l2src_calls.c:
	  When probing the formats and sizes a camera supports, make
	  sure the best ones (highest resolution, prefered format)
	  end up at the beginning of the probed caps and the less
	  desirable ones at the end.  This is important because the
	  order within the caps matters for things like fixation and
	  negotiation, ie. what format is chosen in the end.
	  With recent kernels, the current probing code will end up
	  querying the supported sizes from lowest resolution to
	  highest resolution, adding them to the probed caps in that
	  order, resulting to v4l2src fixating to the lowest possible
	  resolution if downstream does not express a size preference.
	  Also make up a somewhat random ranking of prefered output
	  formats for the same reason. Fixes #485828.
	
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2007-10-11  Tim-Philipp Müller  <tim at centricular dot net>

	Based on patch by: Jason Kivlighn  <jkivlighn gmail com>

	* gst/id3demux/id3v2frames.c:
	  Extract license/copyright URIs from ID3v2 WCOP frames
	  (Fixes #447000).

	* tests/check/elements/id3demux.c:
	* tests/files/Makefile.am:
	* tests/files/id3-447000-wcop.tag:
	  Add simple unit test.

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2007-10-11  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/taglib/gstid3v2mux.cc:
	  Add support for license/copyright URI tags (ID3v2 WCOP frame).
	  Prerequisite for #447000.

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2007-10-08  Jan Schmidt  <Jan.Schmidt@sun.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush):
	Fix compiler warning by using GST_CLOCK_TIME_NONE to initialise
	a GstClockTime.

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

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
	(gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek),
	(gst_rtspsrc_configure_caps), (gst_rtspsrc_loop_udp),
	(gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_play),
	(gst_rtspsrc_change_state):
	More seeking fixes, mostly passing around the new playback segment in
	order to configure it properly.
	Also reset base_time of udp sources when setting them back to PLAYING as
	a temporary hack until core supports seek in live sources properly.

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

	* gst/rtp/gstrtpmp4adepay.c:
	Fix caps as to not confuse autopluggers.

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2007-10-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/id3demux/gstid3demux.c:
	* gst/id3demux/gstid3demux.h:
	* gst/id3demux/id3tags.c:
	* gst/id3demux/id3tags.h:
	* gst/id3demux/id3v2frames.c:
	  Port ID3 tag demuxer over to the new GstTagDemux in -base
	  (now would be a good time to test re-importing your music
	  collection).

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2007-10-06  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/apetag/Makefile.am:
	* gst/apetag/gstapedemux.c:
	* gst/apetag/gstapedemux.h:
	* gst/apetag/gsttagdemux.c:
	* gst/apetag/gsttagdemux.h:
	  Port APE tag demuxer over to the new GstTagDemux in -base.

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2007-10-05  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
	(gst_rtspsrc_perform_seek), (gst_rtspsrc_handle_src_event),
	(gst_rtspsrc_handle_internal_src_query),
	(gst_rtspsrc_handle_src_query), (new_session_pad),
	(gst_rtspsrc_stream_configure_tcp),
	(gst_rtspsrc_stream_configure_transport),
	(gst_rtspsrc_loop_send_cmd):
	Improve flushing behaviour.
	Set state of the udp sources to PAUSE/PLAYING correctly.
	Handle events and queries for UDP and TCP transport now.

Stefan Kost's avatar
Stefan Kost committed
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2007-10-04  Stefan Kost  <ensonic@users.sf.net>

	* gst/rtp/gstrtpgsmdepay.c:
	* gst/rtp/gstrtpgsmpay.c:
	  Add log category.

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

	Patch by: Timo Hotti <Timo.Hotti@sysopendigia.com>

	* tests/check/Makefile.am:
	* tests/check/pipelines/simple-launch-lines.c:
	  Add unit tests for payloaders/depayloaders.

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2007-10-02  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavimux.c:
	* gst/avi/gstavimux.h:
	  Also save codec data for audio streams. Fixes #482495.

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2007-10-02  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavimux.c:
	  Fix "Index entry has invalid stream nr 1".
	  Add support for muxing aac - work in progress (see #482495).

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

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_get_bandwidth),
	(gst_rtspsrc_collect_bandwidth), (gst_rtspsrc_create_stream),
	(gst_rtspsrc_media_to_caps), (gst_rtspsrc_loop_interleaved):
	* gst/rtsp/gstrtspsrc.h:
	Parse bandwidth modifiers, they are not yet configured in the session
	manager because we don't have an API for that yet.

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

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpmap),
	(gst_rtspsrc_media_to_caps), (gst_rtspsrc_loop_interleaved):
	Use shiny new function in -base to get the default clock-rate.
	Update some docs.

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2007-09-29  Sebastien Moutte  <sebastien@moutte.net>

	* win32/MANIFEST:
	Add files to win32 manifest.
	* win32/vs6/libgstaudiofx.dsp:
	* win32/vs6/libgstqtdemux.dsp:
	* win32/vs6/libgstrtp.dsp:
	* win32/vs6/libgstrtsp.dsp:
	Update project files.

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2007-09-28  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_play):
	* gst/rtsp/gstrtspsrc.h:
	In TCP mode, only timestamp the first buffer. TCP is not real time and
	it does not make sense to try to skew compensate, also some servers send
	the first batch of data in a burst.

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2007-09-27  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/matroska/matroska-demux.c:
	  Fix setting the discont flag on the first buffer
	  pushed downstream for formats with private codec
	  data that needs to be deserialised into buffers
	  (such as vorbis and FLAC when in a matroska container).

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2007-09-27  Wim Taymans  <wim.taymans@gmail.com>

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

	* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_init),
	(gst_rtp_mp4v_pay_finalize), (gst_rtp_mp4v_pay_flush),
	(gst_rtp_mp4v_pay_handle_buffer):
	* gst/rtp/gstrtpmp4vpay.h:
	Free the config string. Fixes #480707.
	Clean up the timestamp code a little.

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2007-09-26  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
	(gst_rtspsrc_handle_src_query), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_close):
	* gst/rtsp/gstrtspsrc.h:
	Set timestamps on RTP buffers in interleaved mode.
	Mark first buffers with a DISCONT.
	Remove flush hack now that sync for live sources has been figured out.

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2007-09-26  Wim Taymans  <wim.taymans@gmail.com>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
	Update documentation.

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2007-09-26  Wim Taymans  <wim.taymans@gmail.com>

	* gst/qtdemux/gstrtpxqtdepay.c: (gst_rtp_xqt_depay_process),
	(gst_rtp_xqt_depay_change_state):
	* gst/qtdemux/gstrtpxqtdepay.h:
	Fail if we don't know the quicktime format.

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2007-09-26  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/flac/gstflacenc.c:
	* ext/flac/gstflacenc.h:
	  Save the flow return from the last gst_pad_push() and
	  make sure we pass the right flow return value upstream
	  in the case of failure; minor clean-ups.

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2007-09-25  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/taglib/gstapev2mux.cc:
	* ext/taglib/gstid3v2mux.cc:
	* gst/apetag/gstapedemux.c:
	  Add support for the new GST_TAG_COMPOSER (#459809).

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2007-09-25  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/law/alaw-decode.c:
	* gst/law/alaw-decode.h:
	* gst/law/alaw-encode.c:
	* gst/law/alaw-encode.h:
	* gst/law/alaw.c:
	* gst/law/mulaw-conversion.h:
	  Compulsive clean-ups: use boilerplate macros, add debug
	  categories, fix up things to conform to symbol nomenklatura,
	  etc.

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2007-09-25  Tim-Philipp Müller  <tim at centricular dot net>

	Based on patch by: Laurent Glayal  <spglegle yahoo fr>

	* gst/law/alaw-decode.c:
	* gst/law/alaw-encode.c:
	  Use static tables for A-Law decoding and encoding; this makes
	  A-Law decoding and encoding less CPU-intensive, but increases
	  the binary size a bit. Leaving old code around for now,
	  selectable by a define in the code. Fixes #435435.

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2007-09-25  Sebastian Dröge  <slomo@circular-chaos.org>

	* configure.ac:
	Use AG_GST_ARG_WITH_PLUGINS, AG_GST_ARG_ENABLE_EXTERNAL and
	AG_GST_ARG_ENABLE_EXPERIMENTAL instead of duplicating those macros
	in configure.ac.

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2007-09-25  Sebastian Dröge  <slomo@circular-chaos.org>

	Patch by: <j at bootlab dot org>

	* gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
	Add fourccs for MPEG2 HDV streams. Fixes #479960.

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2007-09-23  Stefan Kost  <ensonic@users.sf.net>

	* sys/oss/gstosshelper.c:
	  Use GST_WARNING instead of a g_critical. This situation is not caused
	  by the application.

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2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	* po/LINGUAS:
	* po/nl.po:
	  Updated translations.

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2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Mikel Olasagasti <hey_neken@mundurat.net>

	* po/eu.po:
	  Added Basque translation.

937 938 939 940 941 942 943 944
2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Abel Cheung <abelcheung@gmail.com>

	* po/zh_HK.po:
	* po/zh_TW.po:
	  Added Chinese (traditional and Hong Kong) translation.

945 946 947 948 949 950 951
2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Jakub Bogusz <qboosh@pld-linux.org>

	* po/pl.po:
	  Added Polish translation.

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2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Ilkka Tuohela <hile@iki.fi>

	* po/fi.po:
	  Added Finnish translation.

959 960 961 962 963 964 965
2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Jorge González González <aloriel@gmail.com>

	* po/es.po:
	  Added Spanish translation.

966 967 968 969 970 971 972
2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Mogens Jaeger <mogens@jaeger.tf>

	* po/da.po:
	  Added Danish translation.

973 974 975 976 977 978 979
2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Funda Wang <fundawang@linux.net.cn>

	* po/zh_CN.po:
	  Added Chinese (simplified) translation.

980 981 982 983 984 985 986
2007-09-22  Thomas Vander Stichele  <thomas at apestaart dot org>

	translated by: Alexander Shopov <ash@contact.bg>

	* po/bg.po:
	  Added Bulgarian translation.

987 988 989 990 991
2007-09-21  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_process):
	Set outgoing packet duration because we can. Fixes #478244 some more.

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2007-09-20  Stefan Kost  <ensonic@users.sf.net>

	* ext/cairo/gsttextoverlay.c:
	  Add info about static leak.
	
	* tests/check/Makefile.am:
	* tests/check/generic/states.c:
	  Improved state change unit test.

Stefan Kost's avatar
Stefan Kost committed
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2007-09-19  Stefan Kost  <ensonic@users.sf.net>

	* docs/plugins/.cvsignore:
	* tests/check/.cvsignore:
	  Ignore registries in any format.

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2007-09-19  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_handle_buffer):
	Removed some unused code.

	* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_handle_buffer):
	* gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_handle_buffer):
	* gst/rtp/gstrtpmp2tpay.c: (gst_rtp_mp2t_pay_handle_buffer):
	* gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_handle_buffer):
	* gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_init_packet),
	(gst_rtp_theora_pay_flush_packet):
	* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_flush_packet):
	Try to preserve the incomming buffer duration on the outgoing
	packets. Fixes #478244.

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2007-09-18  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/taglib/gstapev2mux.cc:
	* ext/taglib/gstid3v2mux.cc:
	  Work around compiler warnings with g++-4.2 when assigning a
	  string constant to a gchar * (partially fixes #478092).

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2007-09-18  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  We require core CVS now for gst_base_src_set_do_timestamp().

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2007-09-17  Jan Schmidt  <Jan.Schmidt@sun.com>

	* gst/audiofx/audiodynamic.c: (gst_audio_dynamic_class_init):
	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
	(new_session_pad), (request_pt_map), (gst_rtspsrc_do_stream_eos),
	(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo),
	(gst_rtspsrc_handle_message):
	Fix compiler warnings shown with Forte.

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2007-09-16  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams),
	(gst_rtspsrc_dup_printf):
	Give meaningfull error when all streams failed to configure for some
	reason.

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2007-09-16  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtp/README:
	Update README with the design for synchronisation rules of RTP on
	sender and receiver.

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2007-09-14  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_loop),
	(gst_wavparse_chain):
	Don't push EOS from the chain function, the element
	driving the pipeline is responsible for this. The bug
	this was meant to fix seems to be queue not forwarding
	EOS in all cases (see #476514).

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2007-09-13  Wim Taymans  <wim.taymans@gmail.com>

	* gst/level/gstlevel.c: (gst_level_class_init), (gst_level_start),
	(gst_level_transform_ip):
	* gst/level/gstlevel.h:
	Use basetransform segment so that it is correctly managed on flushes and
	start/stop.
	Report message timestamp as stream time, which is what an application
	can understand.

Sebastian Dröge's avatar
Sebastian Dröge committed
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2007-09-13  Sebastian Dröge  <slomo@circular-chaos.org>

	* ext/taglib/gstapev2mux.cc:
	* ext/taglib/gstapev2mux.h:
	* ext/taglib/gsttaglibmux.c:
	* tests/check/elements/apev2mux.c:
	Update my mail address.

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2007-09-13  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_eos),
	(gst_wavparse_loop), (gst_wavparse_chain):
	Add EOS logic for the push-based mode too. Fixes #476514.

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2007-09-12  Wim Taymans  <wim.taymans@gmail.com>

	* gst/law/alaw-encode.c: (gst_alawenc_init), (gst_alawenc_chain):
	* gst/law/alaw-encode.h:
	* gst/law/mulaw-encode.c: (gst_mulawenc_init),
	(gst_mulawenc_chain):
	* gst/law/mulaw-encode.h:
	Fix law encoder timestamps.

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2007-09-12  Stefan Kost  <ensonic@users.sf.net>

	* ext/gconf/gstgconfaudiosink.c:
	  Fix warning when building without debug.

	* sys/oss/gstossmixertrack.c:
	  Use const like in alsamixertrack.c (fixes warnings).

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2007-09-11  Wim Taymans  <wim.taymans@gmail.com>

	* sys/v4l2/v4l2src_calls.c:
	(gst_v4l2src_probe_caps_for_format_and_size):
	Fix framerate detection code some more.
	Handle the case where there is a weird step in the stepwise framerates.
	Don't overwrite the min interval with the framerate, use a temp variable
	instead.
	Use max in the Continuous framerate intervals instead of step, which is
	1 according to the docs. Fixes #475424.

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2007-09-10  Wim Taymans  <wim.taymans@gmail.com>

	* gst/udp/gstudpsrc.c: (gst_udpsrc_init), (gst_udpsrc_create):
	Make udpsrc timestamp outgoing buffers based on when they were received.
	Also make it output a segment in time.

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2007-09-10  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c:
	  Plug a little leak. Little code cleanups.

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

	* configure.ac:
	  Use AC_TRY_COMPILE instead of AC_TRY_RUN to check for old
	  flac versions, 's good for cross-compilation karma.

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2007-09-07  Tim-Philipp Müller  <tim at centricular dot net>

	Patch by: Haakon Sporsheim  <haakon.sporsheim at tandberg com>

	* gst/rtp/gstrtph263pay.c:
	  Fix up header structure so that compilers don't add padding
	  between the structure fields, since that would lead to us
	  sending RTP packets with broken headers (as is currently the
	  case when compiling with MSVC). Also see similar fixes in
	  libgstrtp in gst-plugins-base. (#474616; #471194)

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2007-09-07  Wim Taymans  <wim.taymans@gmail.com>

	* sys/v4l2/v4l2src_calls.c:
	(gst_v4l2src_probe_caps_for_format_and_size):
	Don't overwrite our GValue with 0 but instead use the previously
	computed value. Fixes #471823 some more.

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2007-09-06  Tim-Philipp Müller  <tim at centricular dot net>

	* docs/plugins/gst-plugins-good-plugins.hierarchy:
	  No tabs in this file please, or gtk-doc will end up documenting
	  rather absurd class hierarchies.

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2007-09-06  Tim-Philipp Müller  <tim at centricular dot net>

	* ext/gconf/gstswitchsink.c:
	  If the new kid element fails to change state for some reason
	  (e.g. esdsink not being able to connect to the sound server),
	  forward the error message it posted on the bus instead of just
	  posting a generic 'Internal state change error: please file a
	  bug' error message. Fixes #471364.

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2007-09-05  Tim-Philipp Müller  <tim at centricular dot net>

	* gst/qtdemux/Makefile.am:
	* gst/qtdemux/qtdemux.c:
	  Don't assume tags are encoded as UTF-8 (#473670).

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2007-09-05  Tim-Philipp Müller  <tim at centricular dot net>

	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	* sys/v4l2/v4l2src_calls.c:
	  Implement LATENCY queries in the crudest way possible so I don't
	  have to use sync=false any longer when testing with videosinks.

Tim-Philipp Müller's avatar
Tim-Philipp Müller committed
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2007-09-05  Tim-Philipp Müller  <tim at centricular dot net>

	* configure.ac:
	  Fix build.

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2007-09-04  Wim Taymans  <wim.taymans@gmail.com>

	* sys/v4l2/v4l2src_calls.c:
	(gst_v4l2src_probe_caps_for_format_and_size):
	Add some more debugging in the framerate function.
	Iterate stepwise framerate up to and _including_ the max and if nothing
	was added to the list, add a dummy 0/1 to 100/1 framerate so that we
1193
	don't end up with an empty list. Fixes #471823
1194

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2007-09-04  Wim Taymans  <wim.taymans@gmail.com>

	* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
	(gst_multiudpsink_set_clients_string),
	(gst_multiudpsink_get_clients_string),
	(gst_multiudpsink_set_property), (gst_multiudpsink_get_property),
	(gst_multiudpsink_init_send), (gst_multiudpsink_add_internal),
	(gst_multiudpsink_add), (gst_multiudpsink_clear_internal),
	(gst_multiudpsink_clear):
	Add property do configure destination address/port pairs
	API:GstMultiUDPSink::clients

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2007-09-04  Wim Taymans  <wim.taymans@gmail.com>

	* tests/examples/Makefile.am:
	* tests/examples/rtp/Makefile.am:
	* tests/examples/rtp/client-H263p-AMR.sh:
	* tests/examples/rtp/client-H263p-PCMA.sdp:
	* tests/examples/rtp/client-H263p-PCMA.sh:
	* tests/examples/rtp/client-H264-PCMA.sdp:
	* tests/examples/rtp/client-H264-PCMA.sh:
	* tests/examples/rtp/client-PCMA.sh:
	* tests/examples/rtp/server-VTS-H263p-ATS-PCMA.sh:
	* tests/examples/rtp/server-alsasrc-PCMA.sh:
	* tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh:
	* tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh:
	Added some RTP example scripts for sending and receiving RTP streams.

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2007-09-04  Wim Taymans  <wim.taymans@gmail.com>

	* sys/v4l2/gstv4l2src.c: (gst_v4l2_get_caps_info),
	(gst_v4l2src_set_caps), (gst_v4l2src_get_mmap):
	Restructure the setcaps function so that we can also compute the
	expected GStreamer output size of the video frames.
	Set frame_byte_size correctly so that read-based devices have a chance
	of working correctly.
	When grabbing a frame, discard frames that are not of the expected size.
	Some cameras don't output the right framesize for the first buffer.
	Try only a couple of times to get a valid frame, else error out.

	* sys/v4l2/v4l2_calls.c: (gst_v4l2_get_capabilities),
	(gst_v4l2_fill_lists), (gst_v4l2_get_input):
	Add some more debug info when scanning the device.

	* sys/v4l2/v4l2src_calls.c: (gst_v4l2_buffer_new),
	(gst_v4l2_buffer_pool_new), (gst_v4l2_buffer_pool_activate),
	(gst_v4l2src_fill_format_list), (gst_v4l2src_grab_frame),
	(gst_v4l2src_set_capture), (gst_v4l2src_capture_init):
	Add some more debug info when dequeing a frame.

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2007-09-04  Stefan Kost  <ensonic@users.sf.net>

	* gst/wavparse/gstwavparse.c:
	  More code cleanups. Add some more comment and improve debugs logs.

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2007-09-04  Stefan Kost  <ensonic@users.sf.net>

	* gst/wavparse/gstwavparse.c:
	* gst/wavparse/gstwavparse.h:
	  Implement seek-query. Refactor duration calculations. Appropriate use
	  of uint64_scale_int and uint64_scale. Move repeadedly calculated stuff
	  out of loops.

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2007-09-03  Stefan Kost  <ensonic@users.sf.net>

	* gst/avi/gstavidemux.c:
	  Implement seek-query.

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2007-08-29  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink),
	(gst_rtspsrc_dup_printf):
	Use new basesink async property to make sparse RTCP packet not wait for
	preroll.

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2007-08-27  Jan Schmidt  <thaytan@mad.scientist.com>

	* gst/audiofx/Makefile.am:
	Dist the right file.

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2007-08-23  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_dup_printf),
	(gst_rtspsrc_get_float), (gst_rtspsrc_play):
	Make sure we generate and parse floating point values in the POSIX
	locale instead of the current locale. 

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2007-08-22  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_seek),
	(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
	(gst_rtspsrc_play):
	* gst/rtsp/gstrtspsrc.h:
	Fix method detection again.
	Keep track of when we must send a Range header.
	Use segment values for Range, Speed and Scale headers.
	Parse Speed and Scale headers to update the segment values.

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2007-08-22  Stefan Kost  <ensonic@users.sf.net>

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

	* sys/v4l2/v4l2src_calls.c:
	  Handle optional v4l2 ioctls gracefully.

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2007-08-20  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtp/Makefile.am:
	* gst/rtp/gstrtp.c: (plugin_init):
	* gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_base_init),
	(gst_rtp_h263_depay_class_init), (gst_rtp_h263_depay_init),
	(gst_rtp_h263_depay_finalize), (gst_rtp_h263_depay_setcaps),
	(gst_rtp_h263_depay_process), (gst_rtp_h263_depay_set_property),
	(gst_rtp_h263_depay_get_property),
	(gst_rtp_h263_depay_change_state),
	(gst_rtp_h263_depay_plugin_init):
	* gst/rtp/gstrtph263depay.h:
	Added an H263 depayloader. Fixes #369392.

	* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps),
	(gst_rtp_h263p_depay_process):
	* gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type),
	(gst_rtp_h263p_pay_class_init), (gst_rtp_h263p_pay_flush):
	Make the H263+ pay/depayloader support H263-1998 and H263-2000
	payloads.
	Also alow plain H263 on the h263p payloaders. Fixes #465040.

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2007-08-19  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audiofx/audiochebyshevfreqband.c:
	* gst/audiofx/audiochebyshevfreqlimit.c:
	Add small comparision with the windowed sinc filters in the docs.

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2007-08-19  Sebastian Dröge  <slomo@circular-chaos.org>

	* tests/check/elements/audiochebyshevfreqband.c: (GST_START_TEST),
	(audiochebyshevfreqband_suite):
	* tests/check/elements/audiochebyshevfreqlimit.c: (GST_START_TEST),
	(audiochebyshevfreqlimit_suite):
	Also test 32 bit float mode and the type 2 variants of the filters.

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2007-08-18  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
	(gst_rtspsrc_loop):
	Refactor the udp and interleaved loop function a bit.

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2007-08-17  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
	(gst_rtspsrc_finalize), (gst_rtspsrc_connection_send),
	(gst_rtspsrc_connection_receive), (gst_rtspsrc_sink_chain),
	(gst_rtspsrc_handle_request), (gst_rtspsrc_send_keep_alive),
	(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
	(gst_rtspsrc_try_send), (gst_rtspsrc_pause):
	* gst/rtsp/gstrtspsrc.h:
	Protect connection activity with a new lock, avoids deadlocks when going
	to PAUSED. Fixes #455808.

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2007-08-17  Wim Taymans  <wim.taymans@gmail.com>

	* gst/debug/rndbuffersize.c: (gst_rnd_buffer_size_loop):
	Fix debug statement.

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2007-08-17  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_stream_eos):
	Fix stray %u in debug line as spotted by Saur on IRC.

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2007-08-17  Sebastian Dröge  <slomo@circular-chaos.org>

	* gst/audiofx/audiochebyshevfreqband.c:
	(gst_audio_chebyshev_freq_band_class_init):
	* gst/audiofx/audiochebyshevfreqlimit.c:
	(gst_audio_chebyshev_freq_limit_class_init):
	Use generator macros for the process functions for the different
	sample types, add lower upper boundaries for the GObject properties
	so automatically generated UIs can use sliders and add a note about
	the number of poles as a too high number of poles combined with
	very low or very high frequencies will produce only noise.
	* docs/plugins/gst-plugins-good-plugins.args:
	Regenerated for the property changes.

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2007-08-17  Wim Taymans  <wim.taymans@gmail.com>

	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_property),
	(gst_rtspsrc_flush), (gst_rtspsrc_sink_chain),
	(gst_rtspsrc_stream_configure_udp_sink),
	(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_interleaved),
	(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
	(gst_rtspsrc_try_send), (gst_rtspsrc_send),
	(gst_rtspsrc_parse_methods), (gst_rtspsrc_parse_range),
	(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_pause),
	(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
	* gst/rtsp/gstrtspsrc.h:
	Improve timeout handling.
	Use the same socket for sending and receiving RTCP packets so that some
	servers can track clients better.
	Improve connection closed handling. Try to reconnect.
	Don't overwrite our content base with NULL.
	Improve debugging.
	Improve range parsing and handling.
	Remove flushing hack now that core does the right thing.

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2007-08-17  Wim Taymans  <wim.taymans@gmail.com>

	* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
	(gst_multiudpsink_init), (gst_multiudpsink_set_property),
	(gst_multiudpsink_get_property), (gst_multiudpsink_init_send),
	(gst_multiudpsink_close), (gst_multiudpsink_add):
	* gst/udp/gstmultiudpsink.h:
	Add support for getting and setting the socket to use.

	* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
	(gst_udpsrc_create), (gst_udpsrc_get_property):
	Add support for getting the currently used socket.

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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.

1554 1555 1556 1557 1558 1559 1560 1561
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.