Original commit message from CVS:
* gst/level/level-example.c: (main):
Fix for new bus API.
* gst/udp/gstudpsrc.c: (gst_udpsrc_set_property):
Set caps on pads.
Original commit message from CVS:
Updates to payloader/depayloaders, make payloaders use
the base classes.
Updated README with suggested RTP caps and how to convert
to/from SDP.
Added config descriptor in mp4v payloader.
Original commit message from CVS:
2005-09-15 Andy Wingo <wingo@pobox.com>
* gst/autodetect/gstautoaudiosink.c (gst_auto_audio_sink_find_best):
* gst/autodetect/gstautovideosink.c
(gst_auto_video_sink_find_best): Update for new registry API.
Original commit message from CVS:
* common/c-to-xml.py:
* common/gtk-doc-plugins.mak:
a simple py script to generate valid xml from a C example
probably also need to strip an MIT license when we decide
* docs/plugins/Makefile.am:
* gst/level/Makefile.am:
* gst/level/gstlevel.c: (gst_level_init):
* gst/level/level-example.c: (message_handler), (main):
add an example to level that will show up in the docs
* gst/rtp/TODO:
add a note for the future
Original commit message from CVS:
* gst/wavenc/gstwavenc.c: (gst_wavenc_class_init):
Actually define the debug object being used in wavenc. Fixes#316205
Original commit message from CVS:
Link smpte plugin against GST_BASE_LIBS, to get libgstbase; needed to
build on win32 as this plugin uses collectpads (bug 316204)
Original commit message from CVS:
* gst-plugins-good.spec.in:
spec file fixes
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
(gst_multiudpsink_render), (gst_multiudpsink_add),
(gst_multiudpsink_clear):
it actually helps to actually stream if we hook up the
add signal to an actual implementation
* gst/udp/gstudpsrc.c: (gst_udpsrc_start):
some debugging
Original commit message from CVS:
Andrewio Patrickoforus Wingonymus - 5 additional tests for your sins
Add a regression test for level and fix a casting bug that made the additional
channels turn out wrong
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:
* 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:
* ext/mad/Makefile.am:
* gst/avi/Makefile.am:
* gst/effectv/Makefile.am:
* gst/udp/Makefile.am:
* gst/wavparse/Makefile.am:
Use -lgstfoo-@GST_MAJORMINOR@ instead of -lgstfoo-0.9
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers),
(gst_wavparse_stream_data):
Add some fixes from 0.8 branch: allow 24/32bps songs and
blockalign samples to the header-specified size, if any
(#311070); error out on channels==0 or bitrate==0
(#309043, #304588).
