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unsigned char *, not void *, since we do arithm...
Original commit message from CVS:
* gst/audioresample/resample.h:
Declare struct _ResampleState.buffer as unsigned char *, not void *,
since we do arithmetic on it.
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the right places
Original commit message from CVS:
* ext/swfdec/gstswfdec.c: (gst_swfdec_class_init),
(gst_swfdec_chain), (gst_swfdec_render):
Add debugging category and return GstFlowReturn in the right places
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_link):
Get something from the peer pad once we've checked if there is a peer pad.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(qtdemux_tree_get_child_by_type), (qtdemux_parse_trak),
(qtdemux_video_caps):
Couple of fixes
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Original commit message from CVS:
* ext/artsd/gstartsdsink.c: (gst_artsdsink_open_audio):
* gst/festival/gstfestival.c: (socket_receive_file_to_buff):
* gst/vbidec/vbidata.c:
* gst/vbidec/vbidata.h:
* gst/vbidec/vbiscreen.c:
* sys/dxr3/ac3_padder.c:
don't use doc comments for non-docs
change some char* into char[]
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audioresample.
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
Don't leak all input buffers to audioresample.
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is stereo and play it that way instead of ...
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_chanpos_to_gst),
(gst_faad_update_caps):
Assume that an unknown channel mapping with 2 channels
is stereo and play it that way instead of erroring.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(gst_qtdemux_add_stream), (qtdemux_parse_trak):
Handle e.g. jpeg streams with 0 duration frames as having 0 framerate.
Debug fixes. Some 64 bit variable fixes
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Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(gst_qtdemux_add_stream):
Memleak fixes.
Send out EOS for valid reasons (couldn't pull_range() from upstream
for example).
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Original commit message from CVS:
expand tabs
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Original commit message from CVS:
expand tabs
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Original commit message from CVS:
2005-12-05 Andy Wingo <wingo@pobox.com>
* ext/faac/gstfaac.c: (gst_faac_sink_event), (gst_faac_chain):
* ext/faad/gstfaad.c: (gst_faad_chain):
* ext/hermes/gsthermescolorspace.c: (gst_hermes_colorspace_chain):
* ext/lcs/gstcolorspace.c: (gst_colorspace_chain):
* ext/xine/xineinput.c: (gst_xine_input_get):
* gst/colorspace/gstcolorspace.c: (gst_colorspace_chain):
* gst/speed/gstspeed.c: (speed_chain):
* gst/videocrop/gstvideocrop.c: (gst_video_crop_chain): Update for
alloc_buffer changes.
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negotiation etc.. work fine.
Original commit message from CVS:
* gst/audioresample/buffer.c: (audioresample_buffer_queue_flush):
* gst/audioresample/buffer.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
* gst/audioresample/resample.c: (resample_input_flush),
(resample_input_pushthrough), (resample_input_eos),
(resample_get_output_size_for_input),
(resample_get_input_size_for_output), (resample_get_output_size),
(resample_get_output_data):
* gst/audioresample/resample.h:
* gst/audioresample/resample_ref.c: (resample_scale_ref):
Fix audioresample, seek torture, new segments, reverse negotiation
etc.. work fine.
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scalefactor bands exceeded", which result...
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_srcgetcaps):
Handle gracefully the consequence of "Maximum number of scalefactor
bands exceeded", which results in 0 channels with samplerates of 0.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state):
Do upward transitions, then call parent state_change, then do
downward transitions.
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Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(gst_qtdemux_add_stream), (qtdemux_dump_mvhd),
(qtdemux_parse_trak):
Convert to fractional framerates
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(gst_ivorbisfile_loop) gst/qtdemux/qtdemux.c (gst_qtdemu...
Original commit message from CVS:
2005-11-22 Andy Wingo <wingo@pobox.com>
* ext/faad/gstfaad.c (gst_faad_event)
* ext/ivorbis/vorbisfile.c (gst_ivorbisfile_loop)
* gst/qtdemux/qtdemux.c (gst_qtdemux_loop_header)
* gst/speed/gstspeed.c (speed_sink_event)
* gst/tta/gstttaparse.c (gst_tta_parse_src_event)
(gst_tta_parse_parse_header): Run update-funcnames.
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Original commit message from CVS:
* ext/faac/gstfaac.c: (gst_faac_sink_event):
* gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_src_event):
* gst/tta/gstttaparse.c: (gst_tta_parse_src_event):
Fix for stream lock updates.
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Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
(gst_base_audio_sink_provide_clock),
(gst_base_audio_sink_set_clock), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
* gst/audioresample/gstaudioresample.c:
Segment update fix.
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Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
Add DX50, DIVX and DIV3 fourccs (patch by
j@bootlab.org, #321903).
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Original commit message from CVS:
* ext/directfb/dfbvideosink.c:
(gst_dfbvideosink_get_format_from_caps):
* ext/sdl/sdlvideosink.c: (gst_sdlvideosink_create):
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(qtdemux_parse), (qtdemux_type_get), (qtdemux_node_dump_foreach),
(qtdemux_dump_hdlr), (qtdemux_dump_dref), (qtdemux_dump_stsd),
(qtdemux_dump_dcom), (qtdemux_parse_trak), (qtdemux_video_caps),
(qtdemux_audio_caps):
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_v4l2fourcc_to_caps):
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
(gst_v4l2src_capture_init), (gst_v4l2src_get_size_limits):
Update for GST_FOURCC_FORMAT API change.
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Original commit message from CVS:
* ext/audioresample/gstaudioresample.c:
* ext/polyp/polypsink.c: (gst_polypsink_sink_fixate):
* gst/librfb/gstrfbsrc.c: (gst_rfbsrc_fixate):
* gst/modplug/gstmodplug.cc:
* sys/glsink/glimagesink.c: (gst_glimagesink_fixate):
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_fixate):
Rename gst_caps_structure_fixate_* to gst_structure_fixate_*
(#322027)
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Original commit message from CVS:
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
* gst/qtdemux/qtdemux.c: (qtdemux_parse_udta):
Add support for custom genre tags.
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required.
Original commit message from CVS:
* ext/tarkin/wavelet.c:
* ext/tarkin/wavelet.h:
* gst/mpegaudioparse/gstmpegaudioparse.c:
* gst/vbidec/vbidata.c:
* gst/vbidec/vbiscreen.h:
* sys/dxr3/ac3_padder.c:
* sys/dxr3/dxr3audiosink.c:
* sys/dxr3/dxr3spusink.c:
* sys/dxr3/dxr3videosink.c:
* sys/qcam/dark.c:
Don't use gtk-doc markers for normal comments. Fix
gtk-doc formatting where required.
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Original commit message from CVS:
* gst/qtdemux/qtdemux.h:
Remove got_redirect from class structure as well.
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on the bus instead.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_class_init),
(qtdemux_parse_tree):
Remove 'got-redirect' signal and post element message
on the bus instead.
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Original commit message from CVS:
don't put crap in user-visible strings
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Original commit message from CVS:
2005-10-23 Julien MOUTTE <julien@moutte.net>
* gst/tta/gstttaparse.c: (gst_tta_parse_loop): STOPPED->FAILED.
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Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_src_query_types),
(gst_qtdemux_handle_src_query):
* gst/speed/gstspeed.c: (speed_get_query_types), (speed_src_query):
* gst/tta/gstttaparse.c: (gst_tta_parse_src_event),
(gst_tta_parse_get_query_types), (gst_tta_parse_query):
API change fix.
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Original commit message from CVS:
2005-10-18 Julien MOUTTE <julien@moutte.net>
* gst/speed/Makefile.am: Fix build of speed.
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other minor things.
Original commit message from CVS:
Fixed Speed - Recovered featured missed since version 1.37, and changed other
minor things.
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Original commit message from CVS:
restructure configure.ac, use correct libtool LDFLAGS, fix up defines
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enum definition
Original commit message from CVS:
* examples/indexing/indexmpeg.c: (main):
* ext/artsd/gstartsdsink.c: (gst_artsdsink_open_audio),
(gst_artsdsink_close_audio), (gst_artsdsink_change_state):
* ext/artsd/gstartsdsink.h:
* ext/audiofile/gstafparse.c: (gst_afparse_open_file),
(gst_afparse_close_file):
* ext/audiofile/gstafparse.h:
* ext/audiofile/gstafsink.c: (gst_afsink_open_file),
(gst_afsink_close_file), (gst_afsink_chain),
(gst_afsink_change_state):
* ext/audiofile/gstafsink.h:
* ext/audiofile/gstafsrc.c: (gst_afsrc_open_file),
(gst_afsrc_close_file), (gst_afsrc_change_state):
* ext/audiofile/gstafsrc.h:
* ext/cdaudio/gstcdaudio.c: (gst_cdaudio_init):
* ext/directfb/directfbvideosink.c: (gst_directfbvideosink_init):
* ext/dts/gstdtsdec.c: (gst_dtsdec_init):
* ext/jack/gstjack.h:
* ext/jack/gstjackbin.c: (gst_jack_bin_init),
(gst_jack_bin_change_state):
* ext/musepack/gstmusepackdec.c: (gst_musepackdec_init):
* ext/musicbrainz/gsttrm.c: (gst_musicbrainz_init):
* ext/nas/nassink.c: (gst_nassink_open_audio),
(gst_nassink_close_audio), (gst_nassink_change_state):
* ext/nas/nassink.h:
* ext/polyp/polypsink.c: (gst_polypsink_init):
* ext/sdl/sdlvideosink.c: (gst_sdlvideosink_change_state):
* ext/sdl/sdlvideosink.h:
* ext/smoothwave/gstsmoothwave.c: (gst_smoothwave_init):
* ext/sndfile/gstsf.c: (gst_sf_set_property),
(gst_sf_change_state), (gst_sf_release_request_pad),
(gst_sf_open_file), (gst_sf_close_file), (gst_sf_loop):
* ext/sndfile/gstsf.h:
* ext/swfdec/gstswfdec.c: (gst_swfdec_init):
* ext/tarkin/gsttarkindec.c: (gst_tarkindec_init):
* gst/apetag/apedemux.c: (gst_ape_demux_init):
* gst/cdxaparse/gstcdxaparse.c: (gst_cdxaparse_init):
* gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_init):
* gst/festival/gstfestival.c: (gst_festival_change_state):
* gst/festival/gstfestival.h:
* gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_init):
* gst/multifilesink/gstmultifilesink.c: (gst_multifilesink_init),
(gst_multifilesink_set_location), (gst_multifilesink_open_file),
(gst_multifilesink_close_file), (gst_multifilesink_next_file),
(gst_multifilesink_pad_query), (gst_multifilesink_handle_event),
(gst_multifilesink_chain), (gst_multifilesink_change_state):
* gst/multifilesink/gstmultifilesink.h:
* gst/videodrop/gstvideodrop.c: (gst_videodrop_init):
* sys/cdrom/gstcdplayer.c: (cdplayer_init):
* sys/dxr3/dxr3audiosink.c: (dxr3audiosink_init),
(dxr3audiosink_open), (dxr3audiosink_close),
(dxr3audiosink_chain_pcm), (dxr3audiosink_chain_ac3),
(dxr3audiosink_change_state):
* sys/dxr3/dxr3audiosink.h:
* sys/dxr3/dxr3spusink.c: (dxr3spusink_init), (dxr3spusink_open),
(dxr3spusink_close), (dxr3spusink_chain),
(dxr3spusink_change_state):
* sys/dxr3/dxr3spusink.h:
* sys/dxr3/dxr3videosink.c: (dxr3videosink_init),
(dxr3videosink_open), (dxr3videosink_close),
(dxr3videosink_write_data), (dxr3videosink_change_state):
* sys/dxr3/dxr3videosink.h:
* sys/glsink/glimagesink.c: (gst_glimagesink_init):
* sys/qcam/gstqcamsrc.c: (gst_qcamsrc_change_state),
(gst_qcamsrc_open), (gst_qcamsrc_close):
* sys/qcam/gstqcamsrc.h:
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_init):
* sys/vcd/vcdsrc.c: (gst_vcdsrc_set_property), (gst_vcdsrc_get),
(gst_vcdsrc_open_file), (gst_vcdsrc_close_file),
(gst_vcdsrc_change_state), (gst_vcdsrc_recalculate):
* sys/vcd/vcdsrc.h:
renamed GST_FLAGS macros to GST_OBJECT_FLAGS
moved bitshift from macro to enum definition
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Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header):
* gst/tta/gstttaparse.c: (gst_tta_parse_src_event),
(gst_tta_parse_parse_header):
newsegment API update.
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Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header):
No need to take stream lock here.
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Original commit message from CVS:
some disting and build fixes
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Original commit message from CVS:
* gst/audioresample/Makefile.am:
* gst/audioresample/debug.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/resample.c: Convert to using gst debugging
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Original commit message from CVS:
tta plugin ported to 0.9
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Original commit message from CVS:
* check/Makefile.am:
* check/pipelines/simple_launch_lines.c: (setup_pipeline),
(run_pipeline), (GST_START_TEST), (simple_launch_lines_suite):
Add extra tests for basetransform based components.
Comment out the test_element_negotiation test until we decide
if it's testing correct behaviour.
* ext/libvisual/visual.c: (gst_visual_init), (get_buffer),
(gst_visual_chain), (gst_visual_change_state):
Slightly more correct but still bogus timestamping.
Fix state change function.
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init):
* gst/audioresample/gstaudioresample.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_class_init):
* gst/videoscale/gstvideoscale.c: (gst_videoscale_class_init),
(gst_videoscale_prepare_size), (gst_videoscale_set_caps),
(gst_videoscale_prepare_image):
* gst/volume/gstvolume.c: (gst_volume_class_init),
(volume_transform_ip):
Basetransform updates. Enable passthrough modes.
* sys/ximage/ximagesink.c: (gst_ximage_buffer_init),
(gst_ximagesink_renegotiate_size), (gst_ximagesink_xcontext_get),
(gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
Negotiation fix that allows the window to return to the original
size and renegotiate passthrough upstream. Extra debug output.
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Original commit message from CVS:
Ported speed Plugin to GStreamer 0.9
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Original commit message from CVS:
Fix up all the state change functions.
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Original commit message from CVS:
cleaning up bad
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Original commit message from CVS:
* common/gtk-doc-plugins.mak:
* docs/plugins/Makefile.am:
fix distcheck
* gst/audioresample/resample.c:
fix wrong docstring
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Original commit message from CVS:
remove stuff
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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.
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Original commit message from CVS:
2005-08-28 Andy Wingo <wingo@pobox.com>
* Updates for two-arg init from GST_BOILERPLATE_FULL.
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Original commit message from CVS:
use base class' newsegment to properly timestamp
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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
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Original commit message from CVS:
add a check for audioresample
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Original commit message from CVS:
show some info on what's left in the queue
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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
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Original commit message from CVS:
fix broken header setup in Makefile.am
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Original commit message from CVS:
dist more
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Original commit message from CVS:
port audioresample to basetransform
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