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[]
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
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).
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.
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
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.
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.
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.
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
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.
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.
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
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
