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:
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.
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
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.
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)
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.
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.
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...
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.
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.
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...
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.
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.
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...
