=== release 0.10.14 === 2007-08-03 Jan Schmidt <thaytan@mad.scientist.com> * configure.ac: releasing 0.10.14, "Light Years Ahead" 2007-07-27 Jan Schmidt <thaytan@mad.scientist.com> * tests/check/libs/audio.c: (GST_START_TEST): Fix the test to reflect the behaviour of gst_audio_clip_buffer. 2007-07-27 Jan Schmidt <thaytan@mad.scientist.com> * gst-libs/gst/audio/audio.c: When clipping a buffer with no timestamp, assume it is within the segment without warnings. Fixes: #460978 2007-07-27 Wim Taymans <wim.taymans@gmail.com> * gst-libs/gst/rtsp/gstrtspextension.c: (gst_rtsp_extension_send): Fire the signal on the object, not the interface. 2007-07-27 Jan Schmidt <thaytan@mad.scientist.com> * gst-libs/gst/rtsp/.cvsignore: Ber. Don't include the full path, idiot. 2007-07-27 Jan Schmidt <thaytan@mad.scientist.com> * gst-libs/gst/rtsp/.cvsignore: Ignore generated files. 2007-07-26 Jan Schmidt <thaytan@mad.scientist.com> * gst-libs/gst/interfaces/Makefile.am: * gst-libs/gst/interfaces/interfaces-marshal.list: * gst-libs/gst/interfaces/rtspextension.c: * gst-libs/gst/interfaces/rtspextension.h: * gst-libs/gst/rtsp/Makefile.am: * gst-libs/gst/rtsp/gstrtsp.h: * gst-libs/gst/rtsp/gstrtspextension.c: (gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init), (gst_rtsp_extension_detect_server), (gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send), (gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media), (gst_rtsp_extension_configure_stream), (gst_rtsp_extension_get_transports), (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send): * gst-libs/gst/rtsp/gstrtspextension.h: * gst-libs/gst/rtsp/rtsp-marshal.list: Move the rtspextension.h interface into gstrtspextension.h as part of libgstrtsp instead of libgstinterfaces, because it's only for use within plugins, not applications. Add stuff to do the enum & marshal generation needed in libgstrtsp now. Use the GST_TYPE_RTSP_RESULT enum type for the return value of the signal that the GstRTSPExtension interface emits, since G_TYPE_ENUM is abstract. 2007-07-26 Wim Taymans <wim.taymans@gmail.com> * gst-libs/gst/interfaces/Makefile.am: * gst-libs/gst/interfaces/interfaces-marshal.list: * gst-libs/gst/interfaces/rtspextension.c: (gst_rtsp_extension_iface_init), (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send): * gst-libs/gst/interfaces/rtspextension.h: Fix marshaller for the send signal. Add URL to stream selection interface method. 2007-07-26 Jan Schmidt <thaytan@mad.scientist.com> * gst-libs/gst/riff/Makefile.am: Pull in our dependencies from -base before those from outside. 2007-07-26 Wim Taymans <wim.taymans@gmail.com> * docs/libs/gst-plugins-base-libs-sections.txt: * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_decode_ip): * gst-libs/gst/rtsp/gstrtspbase64.h: API: gst_rtsp_base64_decode_ip() Added function to decode Base64 in-place. 2007-07-26 Jan Schmidt <thaytan@mad.scientist.com> * tests/check/libs/.cvsignore: Ignore the mixer test binary. 2007-07-26 Jan Schmidt <thaytan@mad.scientist.com> * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward): Gratuitous comment change to trigger a rebuild on the buildbots. 2007-07-25 Wim Taymans <wim.taymans@gmail.com> * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_media_get_media), (gst_sdp_media_get_port), (gst_sdp_media_get_num_ports), (gst_sdp_media_get_proto), (gst_sdp_media_formats_len), (gst_sdp_media_get_format), (gst_sdp_media_get_information), (gst_sdp_media_connections_len), (gst_sdp_media_get_connection), (gst_sdp_media_bandwidths_len), (gst_sdp_media_get_badwidth), (gst_sdp_media_get_key), (gst_sdp_media_attributes_len), (gst_sdp_media_get_attribute), (gst_sdp_media_get_attribute_val_n), (gst_sdp_media_get_attribute_val): * gst-libs/gst/sdp/gstsdpmessage.h: Constify args where we can. 2007-07-25 Wim Taymans <wim.taymans@gmail.com> * gst-libs/gst/interfaces/Makefile.am: * gst-libs/gst/interfaces/rtspextension.c: (gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init), (gst_rtsp_extension_detect_server), (gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send), (gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media), (gst_rtsp_extension_configure_stream), (gst_rtsp_extension_get_transports), (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send): * gst-libs/gst/interfaces/rtspextension.h: Move interface for RTSP extensions from -good to here. Added helper methods to invoke interface methods. 2007-07-25 Wim Taymans <wim.taymans@gmail.com> * docs/libs/gst-plugins-base-libs-sections.txt: * gst-libs/gst/rtsp/gstrtspdefs.h: * gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach), (gst_rtsp_message_get_type), (gst_rtsp_message_parse_request), (gst_rtsp_message_init_response), (gst_rtsp_message_parse_response), (gst_rtsp_message_new_data), (gst_rtsp_message_parse_data), (gst_rtsp_message_add_header), (gst_rtsp_message_remove_header), (gst_rtsp_message_get_header), (gst_rtsp_message_append_headers), (gst_rtsp_message_set_body), (gst_rtsp_message_get_body), (dump_key_value): * gst-libs/gst/rtsp/gstrtspmessage.h: * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time), (parse_npt_range), (parse_clock_range), (parse_smpte_range), (gst_rtsp_range_parse): * gst-libs/gst/rtsp/gstrtsprange.h: * gst-libs/gst/rtsp/gstrtsptransport.c: * gst-libs/gst/rtsp/gstrtspurl.c: Fix some more RTSP docs. Add some missing methods for dealing with messages. 2007-07-24 Wim Taymans <wim.taymans@gmail.com> * docs/libs/gst-plugins-base-libs-docs.sgml: * docs/libs/gst-plugins-base-libs-sections.txt: * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode): * gst-libs/gst/rtsp/gstrtspbase64.h: * gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_connect), (add_auth_header), (gst_rtsp_connection_write), (gst_rtsp_connection_send), (read_body), (gst_rtsp_connection_receive), (gst_rtsp_connection_next_timeout), (gst_rtsp_connection_reset_timeout), (gst_rtsp_connection_set_auth): * gst-libs/gst/rtsp/gstrtspconnection.h: * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status): * gst-libs/gst/rtsp/gstrtspdefs.h: * gst-libs/gst/rtsp/gstrtspmessage.h: * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time), (parse_npt_range), (parse_clock_range), (parse_smpte_range), (gst_rtsp_range_parse): * gst-libs/gst/rtsp/gstrtspurl.h: Added beginnings of RTSP documentation. 2007-07-24 Wim Taymans <wim.taymans@gmail.com> * docs/libs/Makefile.am: * docs/libs/gst-plugins-base-libs-docs.sgml: * docs/libs/gst-plugins-base-libs-sections.txt: * gst-libs/gst/sdp/gstsdp.h: * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_set_origin), (gst_sdp_message_set_connection), (gst_sdp_message_add_bandwidth), (gst_sdp_message_add_time), (gst_sdp_message_add_zone), (gst_sdp_message_set_key), (gst_sdp_message_get_attribute_val_n), (gst_sdp_message_get_attribute_val), (gst_sdp_message_add_attribute), (gst_sdp_media_new), (gst_sdp_media_init), (gst_sdp_media_uninit), (gst_sdp_media_free), (gst_sdp_media_get_media), (gst_sdp_media_set_media), (gst_sdp_media_get_port), (gst_sdp_media_get_num_ports), (gst_sdp_media_set_port_info), (gst_sdp_media_get_proto), (gst_sdp_media_set_proto), (gst_sdp_media_formats_len), (gst_sdp_media_get_format), (gst_sdp_media_add_format), (gst_sdp_media_get_information), (gst_sdp_media_set_information), (gst_sdp_media_connections_len), (gst_sdp_media_get_connection), (gst_sdp_media_add_connection), (gst_sdp_media_bandwidths_len), (gst_sdp_media_get_badwidth), (gst_sdp_media_add_bandwidth), (gst_sdp_media_set_key), (gst_sdp_media_get_key), (gst_sdp_media_attributes_len), (gst_sdp_media_add_attribute), (gst_sdp_media_get_attribute_val_n), (gst_sdp_media_get_attribute_val), (gst_sdp_message_parse_buffer), (print_media), (gst_sdp_message_dump): * gst-libs/gst/sdp/gstsdpmessage.h: Document the SDP library. Add some of the missing SDPMedia methods. 2007-07-24 Wim Taymans <wim.taymans@gmail.com> * configure.ac: * gst-libs/gst/Makefile.am: * gst-libs/gst/rtsp/Makefile.am: * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode): * gst-libs/gst/rtsp/gstrtspbase64.h: * gst-libs/gst/rtsp/gstrtspconnection.c: (inet_aton), (gst_rtsp_connection_create), (gst_rtsp_connection_connect), (add_auth_header), (add_date_header), (gst_rtsp_connection_write), (gst_rtsp_connection_send), (read_line), (read_string), (read_key), (parse_response_status), (parse_request_line), (parse_line), (gst_rtsp_connection_read), (read_body), (gst_rtsp_connection_receive), (gst_rtsp_connection_close), (gst_rtsp_connection_free), (gst_rtsp_connection_next_timeout), (gst_rtsp_connection_reset_timeout), (gst_rtsp_connection_flush), (gst_rtsp_connection_set_auth): * gst-libs/gst/rtsp/gstrtspconnection.h: * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status), (gst_rtsp_strresult), (gst_rtsp_method_as_text), (gst_rtsp_version_as_text), (gst_rtsp_header_as_text), (gst_rtsp_status_as_text), (gst_rtsp_find_header_field), (gst_rtsp_find_method): * gst-libs/gst/rtsp/gstrtspdefs.h: * gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach), (gst_rtsp_message_new), (gst_rtsp_message_init), (gst_rtsp_message_new_request), (gst_rtsp_message_init_request), (gst_rtsp_message_new_response), (gst_rtsp_message_init_response), (gst_rtsp_message_init_data), (gst_rtsp_message_unset), (gst_rtsp_message_free), (gst_rtsp_message_add_header), (gst_rtsp_message_remove_header), (gst_rtsp_message_get_header), (gst_rtsp_message_append_headers), (gst_rtsp_message_set_body), (gst_rtsp_message_take_body), (gst_rtsp_message_get_body), (gst_rtsp_message_steal_body), (dump_mem), (dump_key_value), (gst_rtsp_message_dump): * gst-libs/gst/rtsp/gstrtspmessage.h: * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time), (parse_npt_range), (parse_clock_range), (parse_smpte_range), (gst_rtsp_range_parse), (gst_rtsp_range_free): * gst-libs/gst/rtsp/gstrtsprange.h: * gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_new), (gst_rtsp_transport_init), (gst_rtsp_transport_get_mime), (gst_rtsp_transport_get_manager), (parse_mode), (parse_range), (range_as_text), (rtsp_transport_mode_as_text), (rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text), (gst_rtsp_transport_parse), (gst_rtsp_transport_as_text), (gst_rtsp_transport_free): * gst-libs/gst/rtsp/gstrtsptransport.h: * gst-libs/gst/rtsp/gstrtspurl.c: (gst_rtsp_url_parse), (gst_rtsp_url_free), (gst_rtsp_url_set_port), (gst_rtsp_url_get_port), (gst_rtsp_url_get_request_uri): * gst-libs/gst/rtsp/gstrtspurl.h: * gst-libs/gst/sdp/Makefile.am: * gst-libs/gst/sdp/gstsdp.h: * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_origin_init), (gst_sdp_connection_init), (gst_sdp_bandwidth_init), (gst_sdp_time_init), (gst_sdp_zone_init), (gst_sdp_key_init), (gst_sdp_attribute_init), (gst_sdp_message_new), (gst_sdp_message_init), (gst_sdp_message_uninit), (gst_sdp_message_free), (gst_sdp_media_new), (gst_sdp_media_init), (gst_sdp_media_uninit), (gst_sdp_media_free), (gst_sdp_message_set_origin), (gst_sdp_message_get_origin), (gst_sdp_message_set_connection), (gst_sdp_message_get_connection), (gst_sdp_message_add_bandwidth), (gst_sdp_message_add_time), (gst_sdp_message_add_zone), (gst_sdp_message_set_key), (gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n), (gst_sdp_message_get_attribute_val), (gst_sdp_message_add_attribute), (gst_sdp_message_add_media), (gst_sdp_media_add_attribute), (gst_sdp_media_add_bandwidth), (gst_sdp_media_add_format), (gst_sdp_media_get_attribute), (gst_sdp_media_get_attribute_val_n), (gst_sdp_media_get_attribute_val), (gst_sdp_media_get_format), (read_string), (read_string_del), (gst_sdp_parse_line), (gst_sdp_message_parse_buffer), (print_media), (gst_sdp_message_dump): * gst-libs/gst/sdp/gstsdpmessage.h: * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in: Move SDP and RTSP from helper objects in -good to a reusable library. Use a proper gst_ namespace. 2007-07-23 Sebastian Dröge <slomo@circular-chaos.org> * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward), (vorbis_dec_flush_decode): Use the new buffer clipping function from gstaudio here. 2007-07-23 Sebastian Dröge <slomo@circular-chaos.org> * docs/libs/gst-plugins-base-libs-sections.txt: * gst-libs/gst/audio/audio.c: (gst_audio_buffer_clip): * gst-libs/gst/audio/audio.h: * tests/check/libs/audio.c: (GST_START_TEST), (audio_suite): API: Add buffer clipping function for raw audio buffers. Fixes #456656. Also add deprecation guards for gst_audio_structure_set_int() to the header. 2007-07-23 Stefan Kost <ensonic@users.sf.net> * docs/libs/gst-plugins-base-libs-sections.txt: Cleanup the docs. 2007-07-23 Wim Taymans <wim.taymans@gmail.com> Patch by: Dan Williams <dcbw at redhat dot com> * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_streaminfo_value_array): Don't return NULL when querying the stream info value array but instead return an empty array. Fixes #459204. 2007-07-23 Tim-Philipp Müller <tim at centricular dot net> * gst/playback/gsturidecodebin.c: Init debug category before using it. 2007-07-21 Jan Schmidt <thaytan@noraisin.net> * gst-libs/gst/interfaces/mixer.h: Add padding vars in place of the signal pointers when building with DISABLE_DEPRECATED so that the interface structure doesn't change size. 2007-07-20 Jan Schmidt <thaytan@noraisin.net> * docs/libs/gst-plugins-base-libs-sections.txt: * ext/alsa/gstalsamixer.c: * ext/alsa/gstalsamixer.h: * ext/alsa/gstalsamixerelement.c: * ext/alsa/gstalsamixertrack.c: * gst-libs/gst/interfaces/mixer.c: * gst-libs/gst/interfaces/mixer.h: * gst-libs/gst/interfaces/mixeroptions.c: * gst-libs/gst/interfaces/mixeroptions.h: * gst-libs/gst/interfaces/mixertrack.c: * gst-libs/gst/interfaces/mixertrack.h: * tests/check/Makefile.am: * tests/check/libs/mixer.c: Patch By: Marc-Andre Lureau <marcandre.lureau@gmail.com> Fixes: #152864 Add support for notifying mixer changes on the message bus, and implement it in alsamixer. API: gst_mixer_get_mixer_flags API: gst_mixer_message_parse_mute_toggled API: gst_mixer_message_parse_record_toggled API: gst_mixer_message_parse_volume_changed API: gst_mixer_message_parse_option_changed API: GstMixerMessageType API: GstMixerFlags 2007-07-20 Michael Smith <msmith@fluendo.com> * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new), (gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps): xcontext->im_format is only for testing XShm support (as the header file comments document). Use xvimage->im_format for everything else. Avoids spurious warnings on buffer allocation before setcaps. 2007-07-20 Stefan Kost <ensonic@users.sf.net> * tests/examples/volume/Makefile.am: * tests/icles/Makefile.am: We should use $(LIBM). 2007-07-20 Stefan Kost <ensonic@users.sf.net> * tests/icles/Makefile.am: This needs -lm. 2007-07-16 Wim Taymans <wim.taymans@gmail.com> * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_class_init), (gst_basertppayload_init), (gst_basertppayload_set_property), (gst_basertppayload_get_property): Don't break ABI, restore previous ranges. Keep the default random selection of timestamp and seqnum offset but as soon as the app sets a specific value, use that one. 2007-07-14 Wim Taymans <wim.taymans@gmail.com> Patch by: Bastien Nocera <hadess at hadess dot net> * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support), (gst_xvimagesink_set_property), (gst_xvimagesink_get_property), (gst_xvimagesink_init), (gst_xvimagesink_class_init): * sys/xvimage/xvimagesink.h: Add option to turn off double-buffering for debugging purposes. Fixes #437169. 2007-07-14 Wim Taymans <wim.taymans@gmail.com> Patch by: Jorn Baayen <jorn at openedhand dot com> * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents), (gst_ximagesink_set_property), (gst_ximagesink_get_property), (gst_ximagesink_init), (gst_ximagesink_class_init): * sys/ximage/ximagesink.h: * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents), (gst_xvimagesink_set_property), (gst_xvimagesink_get_property), (gst_xvimagesink_init), (gst_xvimagesink_class_init): * sys/xvimage/xvimagesink.h: add 'handle-expose' property. Useful for video widgets which may want to be in control of Expose behaviour. Fixes #380625 2007-07-14 Wim Taymans <wim.taymans@gmail.com> * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_class_init), (gst_basertppayload_init), (gst_basertppayload_event), (gst_basertppayload_push), (gst_basertppayload_set_property), (gst_basertppayload_get_property), (gst_basertppayload_change_state): * gst-libs/gst/rtp/gstbasertppayload.h: Fix ranges of rtp payloader properties so that the full range can be used in addition to -1 (random). Fix wrong seqnum reporting in caps. Fixes #420326. 2007-07-13 Wim Taymans <wim.taymans@gmail.com> * gst/videorate/gstvideorate.c: (gst_video_rate_init), (gst_video_rate_query): Use boilerplate. Add latency query, might not be perfect yet but already works a lot better. Fixes #442557. 2007-07-13 Jan Schmidt <thaytan@mad.scientist.com> * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put), (gst_xvimagesink_setcaps): * sys/xvimage/xvimagesink.h: After a caps change, redraw our borders to avoid garbage left there when the image format changes to a smaller size, like 16:9 -> 4:3 Also, hold the flow_lock a bit longer in the set_caps while we're fiddling with the xcontext. 2007-07-13 Jan Schmidt <thaytan@mad.scientist.com> * Makefile.am: * 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. 2007-07-13 Jan Schmidt <thaytan@mad.scientist.com> * gst/ffmpegcolorspace/gstffmpegcodecmap.c: (gst_ffmpeg_caps_to_pixfmt): Fix the r_mask test for RGBA32 on little-endian. Fix a stupid typo that would have obviously broken compilation on big-endian, if anyone was testing. 2007-07-12 Wim Taymans <wim.taymans@gmail.com> * gst/videotestsrc/videotestsrc.c: (paint_hline_AYUV), (paint_hline_str4): * gst/videotestsrc/videotestsrc.h: Add alpha to the color struct. Use a default alpha value of 255 instead of 128. 2007-07-12 Wim Taymans <wim.taymans@gmail.com> * gst/playback/gstplaybasebin.c: (no_more_pads_full), (setup_source): Clear the dynamic pads counter when starting a new uri. This makes reusing playbin work again. Fixes #454264. 2007-07-12 Stefan Kost <ensonic@users.sf.net> * configure.ac: Use pkg-config to locate check. 2007-07-12 Tim-Philipp Müller <tim at centricular dot net> * configure.ac: * tests/check/elements/volume.c: (GST_START_TEST): Fix 'make check' build against core CVS. 2007-07-10 Stefan Kost <ensonic@users.sf.net> * gst-libs/gst/interfaces/propertyprobe.c: * gst-libs/gst/rtp/gstbasertpaudiopayload.c: * gst-libs/gst/tag/gstvorbistag.c: Make gtk-doc happy. 2007-07-08 Tim-Philipp Müller <tim at centricular dot net> * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_callback): Quick hack to make audiosinks stop at EOS when operating in pull-mode; needs to be fixed properly some day. 2007-07-06 Stefan Kost <ensonic@users.sf.net> * docs/libs/gst-plugins-base-libs-sections.txt: Fix location of includes in the docs. 2007-07-06 Jan Schmidt <thaytan@mad.scientist.com> * gst/ffmpegcolorspace/avcodec.h: * gst/ffmpegcolorspace/gstffmpegcodecmap.c: (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt), (gst_ffmpegcsp_avpicture_fill): * gst/ffmpegcolorspace/imgconvert.c: (img_convert), (img_get_alpha_info): Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections of the existing BGRA32 and RGBA32 formats with the alpha at the other end of the word. Partially fixes #451908 2007-07-05 Stefan Kost <ensonic@users.sf.net> * docs/libs/Makefile.am: * docs/plugins/Makefile.am: Simplify --extra-dir as gtkdoc scans recursively. 2007-07-03 Wim Taymans <wim.taymans@gmail.com> * gst/adder/gstadder.c: (gst_adder_sink_getcaps), (gst_adder_request_new_pad): Make getcaps more robust by not using the proxycaps function. This makes sure that we don't end up recursively calling getcaps upstream. See #316248. 2007-06-29 Wim Taymans <wim.taymans@gmail.com> * gst/audioconvert/audioconvert.c: Include math.h to fix compilation. 2007-06-29 Jan Schmidt <thaytan@mad.scientist.com> * gst/ffmpegcolorspace/gstffmpegcodecmap.c: (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt): Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel format, as produced by some dc1394 cameras like the iSight. See http://www.fourcc.org/yuv.php#IYU1 2007-06-28 Sebastian Dröge <slomo@circular-chaos.org> * gst/audioconvert/Makefile.am: * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index), (check_default), (audio_convert_prepare_context), (audio_convert_clean_context), (audio_convert_convert): * gst/audioconvert/audioconvert.h: * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_dithering_get_type), (gst_audio_convert_ns_get_type), (gst_audio_convert_class_init), (gst_audio_convert_init), (gst_audio_convert_set_caps), (gst_audio_convert_set_property), (gst_audio_convert_get_property): * gst/audioconvert/gstaudioconvert.h: * gst/audioconvert/gstaudioquantize.c: (gst_audio_quantize_setup_noise_shaping), (gst_audio_quantize_free_noise_shaping), (gst_audio_quantize_setup_dither), (gst_audio_quantize_free_dither), (gst_audio_quantize_setup_quantize_func), (gst_audio_quantize_setup), (gst_audio_quantize_free): * gst/audioconvert/gstaudioquantize.h: Implement dithering and noise shaping in audioconvert. By default now TPDF dithering (and no noise shaping) will be used when converting from a higher bit depth to 20 bit depth or smaller, otherwise everything will be as it is now. For the last audioconvert in a pipeline it would make sense to use some kind of noise shaping, enabling it by default for all conversions would give undesired results though. Fixes #360246. * tests/check/elements/audioconvert.c: (setup_audioconvert), (GST_START_TEST): Adjust unit test for the new audioconvert. 2007-06-28 Wim Taymans <wim@fluendo.com> * gst/playback/gstqueue2.c: (apply_segment), (update_buffering): Use other metrics as well when estimating the buffer level. 2007-06-28 Wim Taymans <wim@fluendo.com> * gst/playback/gstplaybasebin.c: (make_decoder), (setup_source): Small debug improvement. * gst/playback/gstqueue2.c: (apply_segment), (update_buffering), (plugin_init): Tweak the rate estimation period. When calculating the buffer filledness in rate estimation mode, don't mix it with other metrics. 2007-06-28 Wim Taymans <wim@fluendo.com> * gst/playback/gstdecodebin2.c: (gst_decode_group_new), (gst_decode_group_expose), (gst_decode_group_free), (add_fakesink): When creating the groups, allow for a 5 second, unlimited buffers preroll phase after which we expose the group. When the group is exposed, use a small number of buffers up to a 2 second limit. Also disconnect the overrun signal from multiqueue when we exposed the group because it is not needed anymore. 2007-06-27 Tim-Philipp Müller <tim at centricular dot net> * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8): Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags to utf8-validate; fixes recognition of ID3v1 tags in UTF-8 encoding (#451707); also, output some debugging info when dealing with freeform strings. * tests/check/libs/tag.c: (GST_START_TEST), (tag_suite): Add unit test for the above. 2007-06-27 Tim-Philipp Müller <tim at centricular dot net> * gst-libs/gst/pbutils/descriptions.c: (caps_are_rtp_caps): Add description for Windows Media RTP caps. * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps): Remove RTP fields that don't define the format from caps. 2007-06-27 Tim-Philipp Müller <tim at centricular dot net> * ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer): Skip empty buffers, but not empty header buffers. That way the original vorbisdec unit test still passes (#451145); also, take into account that those empty packets might carry a granulepos. * tests/check/Makefile.am: * tests/check/elements/vorbisdec.c: (_create_codebook_header_buffer), (_create_audio_buffer), (GST_START_TEST), (vorbisdec_suite): Add unit test that sends an empty packet. 2007-06-27 Wim Taymans <wim@fluendo.com> * ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer): Don't error out on 0-sized packets, just emit a warning because this is not a fatal error. Fixes #451145. 2007-06-25 Stefan Kost <ensonic@users.sf.net> * docs/plugins/gst-plugins-base-plugins.args: * docs/plugins/gst-plugins-base-plugins.signals: * docs/plugins/inspect/plugin-adder.xml: * docs/plugins/inspect/plugin-alsa.xml: * docs/plugins/inspect/plugin-audioconvert.xml: * docs/plugins/inspect/plugin-audiorate.xml: * docs/plugins/inspect/plugin-audioresample.xml: * docs/plugins/inspect/plugin-audiotestsrc.xml: * docs/plugins/inspect/plugin-cdparanoia.xml: * docs/plugins/inspect/plugin-decodebin.xml: * docs/plugins/inspect/plugin-decodebin2.xml: * docs/plugins/inspect/plugin-ffmpegcolorspace.xml: * docs/plugins/inspect/plugin-gdp.xml: * docs/plugins/inspect/plugin-gnomevfs.xml: * docs/plugins/inspect/plugin-libvisual.xml: * docs/plugins/inspect/plugin-ogg.xml: * docs/plugins/inspect/plugin-pango.xml: * docs/plugins/inspect/plugin-playbin.xml: * docs/plugins/inspect/plugin-subparse.xml: * docs/plugins/inspect/plugin-tcp.xml: * docs/plugins/inspect/plugin-theora.xml: * docs/plugins/inspect/plugin-typefindfunctions.xml: * docs/plugins/inspect/plugin-video4linux.xml: * docs/plugins/inspect/plugin-videorate.xml: * docs/plugins/inspect/plugin-videoscale.xml: * docs/plugins/inspect/plugin-videotestsrc.xml: * docs/plugins/inspect/plugin-volume.xml: * docs/plugins/inspect/plugin-vorbis.xml: * docs/plugins/inspect/plugin-ximagesink.xml: * docs/plugins/inspect/plugin-xvimagesink.xml: Update docs with caps info. 2007-06-25 Tim-Philipp Müller <tim at centricular dot net> * po/POTFILES.in: Add more files with translatable strings (#450875). 2007-06-23 Edward Hervey <edward@fluendo.com> * ext/ogg/gstoggdemux.c: (gst_ogg_demux_find_chains): The chain should be freed if we error out here, else it will leak. * gst/playback/gstdecodebin.c: (disconnect_unlinked_signals), (cleanup_decodebin): Don't forget to *properly* remove the signals, else it will leak. 2007-06-22 Jan Schmidt <thaytan@noraisin.net> * MAINTAINERS: Updating all the maintainers files 2007-06-21 Stefan Kost <ensonic@users.sf.net> * tests/examples/seek/seek.c: (update_scale), (play_cb), (stop_cb), (main): Destroy and recreate parse-launch based pipeline after stop to be able to play again. Reorder some code and add more comments. 2007-06-20 Jan Schmidt <thaytan@mad.scientist.com> * gst/playback/gstdecodebin2.c: (analyze_new_pad): When handling a delayed-caps notification case, mark the group as dynamic so that the nbdynamic count is incremented and decremented correctly. Fixes: #449156 Patch by: Wim Taymans <wim@fluendo.com> 2007-06-19 Andy Wingo <wingo@pobox.com> * gst-libs/gst/audio/gstbaseaudiosink.c (gst_base_audio_sink_init): Enable pull-mode operation. 2007-06-19 Michael Smith <msmith@fluendo.com> * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps): Change minimum rate back to 1000 to allow low-sample-rate wav files to play back. 2007-06-17 Thomas Vander Stichele <thomas at apestaart dot org> * po/vi.po: Update translations. 2007-06-15 David Schleef <ds@schleef.org> * gst/playback/gstqueue2.c: Fix compile error from ignored return value. 2007-06-15 Michael Smith <msmith@fluendo.com> * gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y): Update tmpbuf for all neccesary rows, not just one, as is required when downscaling. Fixes #402076. 2007-06-15 Michael Smith <msmith@fluendo.com> * tests/check/pipelines/oggmux.c: (validate_ogg_page), (is_video), (eos_buffer_probe): Add a test that ensures we set DELTA_UNIT on all non-header, non-video buffers, if we have a video stream. * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads), (gst_ogg_mux_process_best_pad): Move setting delta_pad to earlier, where we inspect all pads, so that leading audio pages don't get DELTA_UNIT unset if they come before the first DELTA_UNIT from video pages. Fixes the newly-added test. Fixes #385527. 2007-06-14 Tim-Philipp Müller <tim at centricular dot net> * tests/check/pipelines/streamheader.c: (streamheader_suite): Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it fails on the p5-ppc64 build bot and the failure looks like it is due to the same issue as #348114, ie. a compiler bug. 2007-06-13 Edward Hervey <edward@fluendo.com> * gst/playback/gstqueue2.c: (gst_queue_create_read): Fix build on MacOSX. 2007-06-13 Wim Taymans <wim@fluendo.com> * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet), (gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain): Fix compilation on mingw. Fixes #446972. 2007-06-12 Wim Taymans <wim@fluendo.com> Patches by: Thiago Sousa Santos <thiagossantos at gmail dot com> * gst/playback/gstqueue2.c: (update_buffering), (gst_queue_locked_enqueue): Fix a division by zero when the max percent is <= 0. Fixes #446572. also update the buffering status when receiving events. Fixes #446551. 2007-06-11 Wim Taymans <wim@fluendo.com> Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com> * gst/playback/gstqueue2.c: (gst_queue_peer_query), (gst_queue_handle_src_query): Wait for preroll before attempting to forward a duration query upstream. Fixes #445505. 2007-06-07 Sebastien Moutte <sebastien@moutte.net> * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_set_gst_timestamp): Use G_GINT64_CONSTANT macro for int64 constant. * win32/common/libgstinterfaces.def: * win32/common/libgsttag.def: Add new exported functions. 2007-06-07 Tim-Philipp Müller <tim at centricular dot net> * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers): The BOS page of the first Dirac video stream needs to come before the BOS page of any Vorbis streams or other audio streams, just like it is with Theora. 2007-06-07 Wim Taymans <wim@fluendo.com> * gst/playback/gstqueue2.c: (gst_queue_get_range): Fix compilation. 2007-06-06 Wim Taymans <wim@fluendo.com> Patch by: Thiago Sousa Santos <thiagossantos at gmail dot com> * gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_handle_sink_event), (gst_queue_chain), (gst_queue_get_range), (gst_queue_src_checkgetrange_function), (gst_queue_sink_activate_push), (gst_queue_src_activate_push), (gst_queue_src_activate_pull): Add pull based scheduling and fix some deadlocks. Fixes #444523. Does not yet completely work because duration queries upstream won't block yet. 2007-06-06 Wim Taymans <wim@fluendo.com> * configure.ac: * gst/playback/gstqueue2.c: (gst_queue_create_read): Some more fseeko checks. 2007-06-06 Wim Taymans <wim@fluendo.com> * configure.ac: check for large file support. 2007-06-05 Sebastian Dröge <slomo@circular-chaos.org> Based on a patch by Sven Arvidsson <sa at whiz dot se>: * gst/subparse/gstsubparse.c: (parse_subrip), (subviewer_unescape_newlines), (parse_subviewer), (gst_sub_parse_data_format_autodetect), (gst_sub_parse_format_autodetect), (gst_subparse_type_find): * gst/subparse/gstsubparse.h: Add support for SubViewer version 1 and 2 subtitles. Fixes #394061. * tests/check/elements/subparse.c: (GST_START_TEST), (subparse_suite): Add a unit test for both SubViewer formats. 2007-06-01 Michael Smith <msmith@fluendo.com> * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek): Don't overflow intermediate values when seeking to large time values in audiotestsrc. 2007-06-05 Wim Taymans <wim@fluendo.com> * gst/playback/gstqueue2.c: (gst_queue_have_data), (gst_queue_create_read), (gst_queue_read_item_from_file), (gst_queue_open_temp_location_file), (gst_queue_locked_enqueue): Include stdio to define fseeko. 2007-06-05 Wim Taymans <wim@fluendo.com> Patch by: Edward Hervey <edward@fluendo.com> * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_fixate), (gst_v4lsrc_query): Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553. 2007-06-05 Tim-Philipp Müller <tim at centricular dot net> * gst-libs/gst/riff/Makefile.am: * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_info): Use gst_tag_utf8_from_freeform_string() from libgsttag instead of our own implementation. 2007-06-05 Wim Taymans <wim@fluendo.com> * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_set_gst_timestamp), (gst_base_rtp_depayload_change_state): Handle timestamp wraparound. 2007-06-05 Wim Taymans <wim@fluendo.com> * gst/playback/gsturidecodebin.c: (no_more_pads_full), (new_decoded_pad), (remove_pads), (make_decoder), (setup_source), (gst_uri_decode_bin_change_state): Make sure we name srcpads uniquely even when using different internal decodebins. Signal no-more-pads when no more dynamic elements exist. Remove pads on cleanup. 2007-06-05 Wim Taymans <wim@fluendo.com> Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com> * gst/playback/gstqueue2.c: (gst_queue_class_init), (gst_queue_init), (gst_queue_finalize), (gst_queue_write_buffer_to_file), (gst_queue_have_data), (gst_queue_create_read), (gst_queue_read_item_from_file), (gst_queue_open_temp_location_file), (gst_queue_close_temp_location_file), (gst_queue_locked_flush), (gst_queue_locked_enqueue), (gst_queue_locked_dequeue), (gst_queue_is_empty), (gst_queue_is_filled), (gst_queue_change_state), (gst_queue_set_temp_location), (gst_queue_set_property): Add support for filebased buffering. Fixes #441264. 2007-06-05 Wim Taymans <wim@fluendo.com> * gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter), (analyze_new_pad), (connect_pad), (expose_pad), (caps_notify_cb), (caps_notify_group_cb), (gst_decode_group_new), (gst_decode_group_free): Add support for delayed caps fixation when autoplugging. Optimize cases where a multiqueue is not needed/wanted, like right after anything that is not a demuxer. 2007-06-05 Wim Taymans <wim@fluendo.com> * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone), (gst_ogg_pad_submit_packet), (gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain), (gst_ogg_demux_collect_chain_info): consideratly speedup ogg chain detection by not trying to find a base timestamp for skeleton streams. 2007-06-05 Wim Taymans <wim@fluendo.com> * gst/tcp/gstmultifdsink.c: (gst_client_status_get_type), (gst_multi_fd_sink_class_init), (gst_multi_fd_sink_add_full), (gst_multi_fd_sink_remove_flush), (gst_multi_fd_sink_remove_client_link), (gst_multi_fd_sink_handle_client_write), (gst_multi_fd_sink_handle_clients): * gst/tcp/gstmultifdsink.h: Add support for remove_flush. 2007-06-05 Wim Taymans <wim@fluendo.com> * docs/design/draft-keyframe-force.txt: * ext/theora/theoraenc.c: (theora_enc_sink_event), (theora_enc_chain): Add draft design for forcing keyframes in encoders and implement in theoraenc. 2007-06-05 Jan Schmidt <thaytan@mad.scientist.com> * configure.ac: Back to CVS