Original commit message from CVS:
* gst/rtp/TODO:
* gst/rtp/gstrtpmp4vdec.c: (gst_rtpmp4vdec_setcaps):
* gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_class_init),
(gst_rtpmp4venc_init), (gst_rtpmp4venc_parse_data),
(gst_rtpmp4venc_handle_buffer), (gst_rtpmp4venc_set_property),
(gst_rtpmp4venc_get_property):
* gst/rtp/gstrtpmp4venc.h:
Remove g_print.
Update TODO
Make payload encoder a bit smarter and more correct with
timestamps.
Added option in payloader to include config string in-band.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpmap),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_send),
(gst_rtspsrc_change_state):
More SDP parsing and caps setting.
Do NO_PREROLL differently.
add pads only after negotiated.
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
(gst_udpsrc_getcaps):
Implement the getcaps function.
Original commit message from CVS:
* gst/rtp/README:
Update README
* gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_sink_setcaps):
Make extra params as strings.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state),
(gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_send):
Make state change return NO_PREROLL as this is a live
source.
* gst/udp/gstudpsrc.c: (gst_udpsrc_set_property):
Don't unref old caps when NULL.
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).
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.
Original commit message from CVS:
2005-07-08 Andy Wingo <wingo@pobox.com>
* gst/avi/Makefile.am (libgstavi_la_CFLAGS): No gettext hacks, the
defines come from config.h.
* autogen.sh: Run autopoint, etc.
* Makefile.am (DIST_SUBDIRS, SUBDIRS): Go into po/.
* configure.ac: Add gettext stuff.
Original commit message from CVS:
* gst/videobox/gstvideobox.c: (gst_video_box_init),
(gst_video_box_transform_caps), (gst_video_box_set_caps):
Logic was reversed. Needs some more fixes in the transform
function to include AYUV output.
Moved AYUV as prefered format.
Original commit message from CVS:
* gst/base/gstbasesrc.c: (gst_base_src_get_range),
(gst_base_src_default_negotiate), (gst_base_src_negotiate):
Allow subclasses to implement their own negotiation.
Original commit message from CVS:
2005-07-05 Andy Wingo <wingo@pobox.com>
* gst/videobox/gstvideobox.c: Clean up, port to 0.9, use
BaseTransform.
* gst/videobox/Makefile.am: Link to base libs, include
plugins-base cflags, dist the README.
* configure.ac (GST_PLUGIN_ALL, AC_CONFIG_FILES): Add videobox to
the build.
Original commit message from CVS:
2005-07-04 Andy Wingo <wingo@pobox.com>
* examples/level/:
* examples/level/Makefile.am:
* examples/level/README:
* examples/level/demo.c:
* examples/level/plot.c: Examples moved out of the source dir. Not
updated tho.
* configure.ac: Add level to the build.
* gst/level/Makefile.am:
* gst/level/gstlevel.h:
* gst/level/gstlevel.c: Cleaned up, ported to 0.9.
Original commit message from CVS:
* gst/udp/Makefile.am:
* gst/udp/gstudp.c:
* gst/udp/gstdynudpsink.c: (new)
* gst/udp/gstdynudpsink.h: (new)
Added new element (udpdynsink) that receives GstNetBuffers and sends the
udp packets to the source given in the buffer. It's used by rtpsession
element for now.
* gst/udp/gstudpsrc.c:
Fixed memory leak.
Original commit message from CVS:
2005-07-01 Jan Schmidt <thaytan@mad.scientist.com>
* ext/libcaca/Makefile.am:
* ext/mad/Makefile.am:
* gst/effectv/Makefile.am:
* gst/udp/Makefile.am:
Replace GST_PLUGINS_LIBS_* with GST_PLUGINS_BASE_*
* ext/mad/gstid3tag.c: (gst_id3_tag_src_query),
(gst_id3_tag_src_event), (gst_id3_tag_sink_event),
(gst_id3_tag_chain), (plugin_init):
* ext/mad/gstmad.c: (gst_mad_src_query), (gst_mad_chain):
Signedness warning fix, use gst_pad_get_peer instead of GST_PAD_PEER
in querying and event handling, because we're not holding the pad
lock and the peer may disappear.
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_subindex),
(gst_avi_demux_parse_index), (gst_avi_demux_massage_index):
Signedness warning fixes.
* gst/videofilter/gstvideotemplate.c: (plugin_init):
Remove gst_library_load
Original commit message from CVS:
* gst/avi/Makefile.am: (libgstavi_la_LIBADD):
Added linking to libgstriff-0.9
* ext/mad/gstmad.c: (gst_mad_src_query):
check the format of the upstream query and return query if it's the
same format as the requested one.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_create_stream),
(gst_rtspsrc_add_element), (gst_rtspsrc_set_state),
(gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (find_stream),
(gst_rtspsrc_loop), (gst_rtspsrc_open), (gst_rtspsrc_play),
(gst_rtspsrc_change_state):
Fix case where outpad could not be decided.
