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2005-08-30Make autogen work again.Jan Schmidt3-5/+8
Original commit message from CVS: * autogen.sh: * configure.ac: Make autogen work again.
2005-08-30check/elements/: Fix checks.Wim Taymans2-0/+2
Original commit message from CVS: * check/elements/audioconvert.c: (setup_audioconvert): * check/elements/audioresample.c: (setup_audioresample): * check/elements/volume.c: (setup_volume): Fix checks.
2005-08-30remove stuffThomas Vander Stichele16-3413/+0
Original commit message from CVS: remove stuff
2005-08-30all these plugins are moved to gst-plugins-goodThomas Vander Stichele4-359/+175
Original commit message from CVS: all these plugins are moved to gst-plugins-good
2005-08-30Ported to GStreamer 0.9. Need to fix performance issues.Flavio Oliveira3-0/+7
Original commit message from CVS: Ported to GStreamer 0.9. Need to fix performance issues.
2005-08-28Updates for two-arg init from GST_BOILERPLATE.Andy Wingo23-1420/+905
Original commit message from CVS: 2005-08-28 Andy Wingo <wingo@pobox.com> * Updates for two-arg init from GST_BOILERPLATE. * ext/ladspa/gstsignalprocessor.c (gst_signal_processor_init): Use the second arg for the class, because G_OBJECT_GET_CLASS (self) returns the wrong thing. (gst_signal_processor_add_pad_from_template): Make pads of the right type. * ext/ladspa/gstladspa.c (gst_ladspa_class_get_param_spec): Make writable param specs G_PARAM_CONSTRUCT so default values work. (gst_ladspa_init): Use the second arg for the class.
2005-08-28Updates for two-arg init from GST_BOILERPLATE_FULL.Andy Wingo1-4/+2
Original commit message from CVS: 2005-08-28 Andy Wingo <wingo@pobox.com> * Updates for two-arg init from GST_BOILERPLATE_FULL.
2005-08-26use base class' newsegment to properly timestampThomas Vander Stichele1-2/+5
Original commit message from CVS: use base class' newsegment to properly timestamp
2005-08-26ext/ladspa/gstladspa.*: Finish porting, still doesn't work but it does ↵Andy Wingo6-678/+333
compile and register. I have more features tha... Original commit message from CVS: 2005-08-26 Andy Wingo <wingo@pobox.com> * ext/ladspa/gstladspa.c: * ext/ladspa/gstladspa.h: Finish porting, still doesn't work but it does compile and register. I have more features than you. * ext/ladspa/gstsignalprocessor.h: * ext/ladspa/gstsignalprocessor.c: Updates, bug fixen.
2005-08-26do proper cleanup/creation, fixes state changesThomas Vander Stichele1-0/+6
Original commit message from CVS: do proper cleanup/creation, fixes state changes
2005-08-25check/: add a test for audioconvertThomas Vander Stichele2-4/+10
Original commit message from CVS: * check/Makefile.am: * check/elements/audioconvert.c: (setup_audioconvert), (cleanup_audioconvert), (get_int_caps), (verify_convert), (GST_START_TEST), (audioconvert_suite), (main): add a test for audioconvert * gst/audioresample/gstaudioresample.c: * gst/audioresample/gstaudioresample.h: set DURATION so that TIMESTAMP(a) + DURATION(a) == TIMESTAMP(b); note that for buffers of 1/3 sec this means DURATION(c) is one nanosecond more than for a and b
2005-08-25some more testing for perfect streamsThomas Vander Stichele1-6/+23
Original commit message from CVS: some more testing for perfect streams
2005-08-25add a check for audioresampleThomas Vander Stichele3-2/+240
Original commit message from CVS: add a check for audioresample
2005-08-25show some info on what's left in the queueThomas Vander Stichele1-2/+6
Original commit message from CVS: show some info on what's left in the queue
2005-08-25gst/audioresample/: add room for extra overlap samples when asked to ↵Thomas Vander Stichele6-35/+73
transform size protect against possible mem corr... Original commit message from CVS: * gst/audioresample/debug.c: * gst/audioresample/gstaudioresample.c: add room for extra overlap samples when asked to transform size protect against possible mem corruption and check for discrepancies between written size and outbuffer's size so we can warn for potential problems * gst/audioresample/resample.c: (resample_init), (resample_get_output_size_for_input), (resample_get_output_size), (resample_set_n_channels), (resample_set_format): set debug level based on RESAMPLE_DEBUG env var make sure that get_output_size* returns a whole number of sample_size set sample_size each time either channel or format is set * gst/audioresample/resample_chunk.c: (resample_scale_chunk): * gst/audioresample/resample_functable.c: (resample_scale_functable): * gst/audioresample/resample_ref.c: (resample_scale_ref): remove r->sample_size, it's done in resample.c now add some debugging to the ref implementation make sure we only give back bytes that are wholes of the sample size
2005-08-25gst/level/gstlevel.c: Revert unpopular change for GST_MESSAGE_SRC to GObject.Jan Schmidt1-0/+5
Original commit message from CVS: * gst/level/gstlevel.c: (gst_level_message_new): Revert unpopular change for GST_MESSAGE_SRC to GObject.
2005-08-25add missing filesAndy Wingo2-0/+799
Original commit message from CVS: add missing files
2005-08-25ext/ladspa/gstladspa.*: Halfway-ported. Doesn't compile yet.Andy Wingo6-143/+180
Original commit message from CVS: 2005-08-25 Andy Wingo <wingo@pobox.com> * ext/ladspa/gstladspa.h: * ext/ladspa/gstladspa.c: Halfway-ported. Doesn't compile yet. * ext/ladspa/gstsignalprocessor.h: * ext/ladspa/gstsignalprocessor.c: New files, the start of a base class for DSP elements. * configure.ac: Sort the external libs checks, add a ladspa check, output the ladspa makefile.
2005-08-25Fixed EOS and improved robustness for malformed indices.Owen Fraser-Green1-0/+7
Original commit message from CVS: Fixed EOS and improved robustness for malformed indices.
2005-08-24add lameChristian Schaller1-3/+3
Original commit message from CVS: add lame
2005-08-24fix broken header setup in Makefile.amChristian Schaller1-1/+1
Original commit message from CVS: fix broken header setup in Makefile.am
2005-08-24dist moreThomas Vander Stichele2-5/+6
Original commit message from CVS: dist more
2005-08-24ext/dv/gstdvdemux.c (gst_dvdemux_demux_frame): Send out valid segment end ↵Andy Wingo1-0/+10
timestamps. Original commit message from CVS: 2005-08-24 Andy Wingo <wingo@pobox.com> * ext/dv/gstdvdemux.c (gst_dvdemux_demux_frame): Send out valid segment end timestamps. (Also commit an old changelog entry)
2005-08-24port audioresample to basetransformThomas Vander Stichele10-236/+271
Original commit message from CVS: port audioresample to basetransform
2005-08-24enable more; update for basetransformThomas Vander Stichele3-5/+15
Original commit message from CVS: enable more; update for basetransform
2005-08-24gst/level/gstlevel.c: GST_MESSAGE_SRC became a GObjectJan Schmidt1-0/+4
Original commit message from CVS: * gst/level/gstlevel.c: (gst_level_message_new): GST_MESSAGE_SRC became a GObject
2005-08-24fix speex entryChristian Schaller1-1/+1
Original commit message from CVS: fix speex entry
2005-08-24add speex to spec file and remove gstosslement from POTFILES.inChristian Schaller2-7/+6
Original commit message from CVS: add speex to spec file and remove gstosslement from POTFILES.in
2005-08-23ext/speex/gstspeexenc.h: Fixed include path of adapterStefan Kost1-0/+5
Original commit message from CVS: * ext/speex/gstspeexenc.h: Fixed include path of adapter
2005-08-23gst/audioresample/Makefile.am: Leet audioresampling codeDavid Schleef14-0/+2254
Original commit message from CVS: * gst/audioresample/Makefile.am: Leet audioresampling code * gst/audioresample/buffer.c: * gst/audioresample/buffer.h: * gst/audioresample/debug.c: * gst/audioresample/debug.h: * gst/audioresample/functable.c: * gst/audioresample/functable.h: * gst/audioresample/gstaudioresample.c: * gst/audioresample/gstaudioresample.h: * gst/audioresample/resample.c: * gst/audioresample/resample.h: * gst/audioresample/resample_chunk.c: * gst/audioresample/resample_functable.c: * gst/audioresample/resample_ref.c:
2005-08-23ext/speex/: Fix property warning.Wim Taymans1-0/+6
Original commit message from CVS: * ext/speex/gstspeexdec.c: (gst_speex_dec_class_init): * ext/speex/gstspeexenc.c: (gst_speexenc_class_init): Fix property warning.
2005-08-23gst/rtp/: Small updates, RFC reference to payload encoders.Wim Taymans1-0/+10
Original commit message from CVS: * gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_init), (gst_rtpamrdec_sink_setcaps), (gst_rtpamrdec_chain): * gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_class_init), (gst_rtpamrenc_init), (gst_rtpamrenc_chain): * gst/rtp/gstrtph263penc.c: (gst_rtph263penc_class_init), (gst_rtph263penc_flush), (gst_rtph263penc_chain): Small updates, RFC reference to payload encoders.
2005-08-23Port speexdec. Leads to some unfamiliar warnings on console, but works ↵Ronald S. Bultje3-6/+63
otherwise. Original commit message from CVS: * configure.ac: * ext/Makefile.am: * ext/speex/Makefile.am: * ext/speex/gstspeex.c: (plugin_init): * ext/speex/gstspeexdec.c: (speex_get_query_types), (gst_speex_dec_init), (speex_dec_src_query), (speex_dec_src_event), (speex_dec_event), (speex_dec_chain): Port speexdec. Leads to some unfamiliar warnings on console, but works otherwise.
2005-08-23sys/oss/gstosssrc.c (gst_oss_src_open): Set the device-name property after ↵Andy Wingo1-0/+3
opening the mixer. Original commit message from CVS: 2005-08-23 Andy Wingo <wingo@pobox.com> * sys/oss/gstosssrc.c (gst_oss_src_open): Set the device-name property after opening the mixer.
2005-08-23sys/oss/gstosssrc.*: Easy to implement a mixer, eh...Andy Wingo1-0/+3
Original commit message from CVS: 2005-08-23 Andy Wingo <wingo@pobox.com> * sys/oss/gstosssrc.c: * sys/oss/gstosssrc.h: Easy to implement a mixer, eh...
2005-08-23sys/oss/gstossmixerelement.*: Added mixer element like alsamixer.Andy Wingo1-0/+7
Original commit message from CVS: 2005-08-23 Andy Wingo <wingo@pobox.com> * sys/oss/gstossmixerelement.h: * sys/oss/gstossmixerelement.c: Added mixer element like alsamixer. * sys/oss/Makefile.am: * sys/oss/gstossaudio.c: Register the ossmixer element.
2005-08-23changelogAndy Wingo1-0/+24
Original commit message from CVS: changelog
2005-08-23sys/oss/gstosssrc.*: Totally ported, dude.Andy Wingo1-0/+10
Original commit message from CVS: 2005-08-23 Andy Wingo <wingo@pobox.com> * sys/oss/gstosssrc.h: * sys/oss/gstosssrc.c: Totally ported, dude. * sys/oss/Makefile.am: * sys/oss/gstossaudio.c: Add osssrc. * sys/oss/gstosssink.c: We do native byte order.
2005-08-23Fixed mishandling events and incorrect audio skipping after seek.Owen Fraser-Green1-0/+6
Original commit message from CVS: Fixed mishandling events and incorrect audio skipping after seek.
2005-08-22ext/mad/gstid3tag.c: Works a bit better now, but still needs a rewrite to ↵Jan Schmidt1-0/+8
use get_range instead of this seeking nasti... Original commit message from CVS: * ext/mad/gstid3tag.c: (gst_id3_tag_init), (gst_id3_tag_sink_event), (gst_id3_tag_do_caps_nego), (gst_id3_tag_chain), (gst_id3_tag_change_state), (plugin_init): Works a bit better now, but still needs a rewrite to use get_range instead of this seeking nastiness.
2005-08-22Port flacdec (seeking is still slow'ish).Ronald S. Bultje3-4/+36
Original commit message from CVS: * configure.ac: * ext/Makefile.am: * ext/flac/Makefile.am: * ext/flac/gstflac.c: (plugin_init): * ext/flac/gstflacdec.c: (flacdec_get_type), (gst_flacdec_init), (gst_flacdec_update_metadata), (gst_flacdec_seek), (gst_flacdec_tell), (gst_flacdec_length), (gst_flacdec_read), (gst_flacdec_write), (gst_flacdec_loop), (gst_flacdec_get_src_query_types), (gst_flacdec_src_query), (gst_flacdec_src_event), (gst_flacdec_sink_activate), (gst_flacdec_sink_activate_pull), (gst_flacdec_change_state): * ext/flac/gstflacdec.h: Port flacdec (seeking is still slow'ish).
2005-08-22Fixed some seeking issuesOwen Fraser-Green2-0/+6
Original commit message from CVS: Fixed some seeking issues
2005-08-19add mpegaudioparse to spec fileChristian Schaller1-0/+1
Original commit message from CVS: add mpegaudioparse to spec file
2005-08-19gst/udp/gstmultiudpsink.c: Remove get_time code that is both wrong and unneeded.Wim Taymans1-0/+5
Original commit message from CVS: * gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init): Remove get_time code that is both wrong and unneeded.
2005-08-19gst/rtp/gstrtph263penc.*: Added configurable pt and ssrc, to be merged in ↵Wim Taymans1-0/+9
the caps or a base class... Original commit message from CVS: * gst/rtp/gstrtph263penc.c: (gst_rtph263penc_class_init), (gst_rtph263penc_flush), (gst_rtph263penc_chain), (gst_rtph263penc_set_property), (gst_rtph263penc_get_property): * gst/rtp/gstrtph263penc.h: Added configurable pt and ssrc, to be merged in the caps or a base class...
2005-08-19gst/rtp/: Some cleanups in the h263p (de)payloaders.Wim Taymans1-0/+8
Original commit message from CVS: * gst/rtp/gstrtph263pdec.c: (gst_rtph263pdec_init), (gst_rtph263pdec_chain): * gst/rtp/gstrtph263penc.c: (gst_rtph263penc_class_init), (gst_rtph263penc_flush), (gst_rtph263penc_chain): Some cleanups in the h263p (de)payloaders.
2005-08-19add mpegaudioparse to configure.acChristian Schaller1-0/+1
Original commit message from CVS: add mpegaudioparse to configure.ac
2005-08-19latest makefile and spec file fixesChristian Schaller2-2/+4
Original commit message from CVS: latest makefile and spec file fixes
2005-08-19ext/amrnb/: Update caps with audio/AMR.Wim Taymans1-0/+27
Original commit message from CVS: * ext/amrnb/amrnbdec.c: * ext/amrnb/amrnbenc.c: (gst_amrnbenc_setcaps): * ext/amrnb/amrnbparse.c: Update caps with audio/AMR. * gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_init), (gst_rtpamrdec_sink_setcaps), (gst_rtpamrdec_chain), (gst_rtpamrdec_change_state): * gst/rtp/gstrtpamrdec.h: * gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_class_init), (gst_rtpamrenc_init), (gst_rtpamrenc_chain): Dont set FT headers twice, it was already in the encoded bitstream. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play): * gst/rtsp/rtspconnection.c: (parse_line): Cleanups * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_create), (gst_udpsrc_set_property), (gst_udpsrc_get_property): * gst/udp/gstudpsrc.h: Added caps property, we need this soon to type the buffers.
2005-08-18gst/rtp/gstrtpamrdec.c: Fix up amr depayloader a bit.Wim Taymans1-0/+11
Original commit message from CVS: * gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_init), (gst_rtpamrdec_chain): Fix up amr depayloader a bit. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play): Look for options result in Public and Allow header fields.. spec says Allow but some servers return Public...
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