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).
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_class_init), (gst_faad_setcaps):
Add debug category, remove Close() call that made it crash
whenever reusing, renegotiating or anything; Close() actually
free()s the handle and should only be called on READY->NULL.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header):
Actually set caps on buffer (in addition to pad), also.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_header):
Fix AVI header parsing: add missing break statement after
GST_RIFF_INFO_LIST parsing code; gst_riff_read_chunk() has
already advanced the avi->offset, no need to do it twice
(fixes MovieOfMovies.avi).
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_event),
(gst_avi_demux_handle_seek):
Fix seeking (or, well, fix threading issue where a variable was
set before a lock was taken and was already unset before that
same lock was taken and was thus no longer in existance when it
actually had to be used).
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry):
Mixing binary and logical operators is not going to work; fix
position-querying in Totem.
Original commit message from CVS:
2005-08-04 Andy Wingo <wingo@pobox.com>
* sys/oss/gstossaudio.c (plugin_init): Second-class citizen.
* gst/videobox/gstvideobox.c (gst_video_box_get_size): Update for
API changes.
* configure.ac (DEFAULT_AUDIOSINK, DEFAULT_VIDEOSINK): Set to
autoaudiosink and autovideosink.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_parse_stream), (gst_avi_demux_process_next_entry):
You need to allocatate (len+1) characters to store a len size string.
Also don't stop the processing task if the output pad is not linked.
Original commit message from CVS:
* configure.ac:
* gst/wavparse/gstwavparse.c:
* gst/wavparse/gstwavparse.h:
* gst/wavparse/Makefile.am:
Ported wavparse to 0.9 . Playing, seeking and state changes work.
Could need more loving on the headers though.
Original commit message from CVS:
2005-07-16 Philippe Khalaf <burger@speedy.org>
* gst/fdsrc/gstfdsrc.c:
* gst/fdsrc/gstfdsrc.h:
* gst/fdsrc/Makefile.am:
Moved fdsrc 0.9 port from gstreamer/gst/elements to here.
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_transform):
* gst/videobox/gstvideobox.c: (gst_video_box_class_init),
(gst_video_box_get_size), (gst_video_box_transform):
Port to new base class.
