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:
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:
2005-08-23 Andy Wingo <wingo@pobox.com>
* sys/oss/gstossmixer.h:
* sys/oss/gstossmixer.c: Refactored to be more like alsamixer.
* sys/oss/gstossmixertrack.h:
* sys/oss/gstossmixertrack.c: Split out from gstossmixer.[ch],
like gstalsamixer.
* sys/oss/gstosssrc.c:
* sys/oss/gstosssink.c: Where before we used a gstosselement
object as a helper library, now just call functions from
gstosshelper.
* sys/oss/gstosshelper.h:
* sys/oss/gstosshelper.c: Made a real library. Removed
propertyprobe for now, should add it back later.
* sys/oss/gstosselement.h:
* sys/oss/gstosselement.c: Removed, we don't have a shared base
class.
* sys/oss/gstosshelper.c (gst_oss_helper_probe_caps): Search
higher-to-lower, makes 16 bit appear earlier in the caps, which
makes it preferred.
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...
Original commit message from CVS:
* gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_class_init),
(gst_rtpamrenc_init), (gst_rtpamrenc_chain),
(gst_rtpamrenc_set_property), (gst_rtpamrenc_get_property):
* gst/rtp/gstrtpamrenc.h:
Added payload_type and ssrc properties to the payloader.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play):
Options need to be stripped and are in the Public header field.
* gst/rtsp/rtspurl.c: (rtsp_url_parse):
Fix url / parsing...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
(gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspdefs.c: (rtsp_method_as_text), (rtsp_find_method):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtsptransport.c: (rtsp_transport_parse):
Handle RTSP defaults better.
Issue OPTIONS request to figure out what we are allowed to do.
Make the methods a bitfield so we can easily collect supported
options.
Fix rtsp_find_method.
Do proper RTSP connection shutdown.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
(gst_avi_demux_stream_header):
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(gst_qtdemux_add_stream), (qtdemux_parse_tree):
Uncomment metadata and codec-name handling.
Original commit message from CVS:
2005-08-16 Andy Wingo <wingo@pobox.com>
* ext/raw1394/gstdv1394src.c (gst_dv1394src_iso_receive): Note
license info in the source code -- was only in the commit log
before.
* ext/dv/gstdvdec.h:
* ext/dv/gstdvdec.c: Only decodes systemstream=FALSE dv video --
old pipelines using dvdec should probably have a dvdemux first.
* ext/dv/gstdvdemux.h:
* ext/dv/gstdvdemux.c: Split out from dvdec, chunks the incoming
systemstream=TRUE data into frames, sets caps data, and spits out
PCM audio in addition to systemstream=FALSE video frames. Operates
in chain mode only for now; should make a getrange version as
well.
* ext/dv/gstdv.c: New file, registers the libgstdv plugin.
* ext/dv/Makefile.am: Library name changed to libgstdv. Split
dvdec into dvdemux and dvdec.
