Original commit message from CVS:
* ext/Makefile.am:
* ext/cairo/Makefile.am:
* ext/cairo/gstcairo.c: (plugin_init):
* ext/cairo/gsttextoverlay.c: (gst_textoverlay_change_state):
* ext/cairo/gsttimeoverlay.c: (gst_timeoverlay_update_font_height),
(gst_timeoverlay_setup), (gst_timeoverlay_planar411):
* ext/cairo/gsttimeoverlay.h:
update of cairo-based timeoverlay to 1.0 Cairo API
doesn't work yet for resizing of output sink
Original commit message from CVS:
* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_reset):
* ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_reset):
Make sure element is NULL before removing from the bin.
Original commit message from CVS:
(gst_dv1394src_bus_reset): Post a message when the cable is
unplugged.
(gst_dv1394src_create, gst_dv1394src_unlock): Remove some prints.
Original commit message from CVS:
2005-10-07 Andy Wingo <wingo@pobox.com>
* ext/raw1394/gstdv1394src.c: Make interruptible, so it won't
block forever in a read().
Original commit message from CVS:
2005-10-07 Andy Wingo <wingo@pobox.com>
* ext/raw1394/gstdv1394src.c: Clean up for style before doing some
hacking. The only change should be that the state change stuff was
put into basesrc's start() and stop() routines, which coalesces
some steps.
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_reset),
(gst_auto_audio_sink_find_best), (gst_auto_audio_sink_detect):
* gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_reset),
(gst_auto_video_sink_find_best), (gst_auto_video_sink_detect):
Set state of elements to NULL before removing from bins.
Set state of test element to NULL if we failed to move it to READY
Original commit message from CVS:
* ext/dv/Makefile.am:
* ext/dv/gstdvdemux.c: (gst_dvdemux_src_query), (gst_dvdemux_src_conver):
Added DEFAULT <==> BYTES, TIME conversions on srcpad,
Corrected the query function for position so it doesn't forget what
format was asked, and calls the conversion functions on the correct pad.
Original commit message from CVS:
2005-10-03 Andy Wingo <wingo@pobox.com>
* ext/flac/gstflacdec.c (gst_flacdec_write): Deal with pad_alloc
error returns.
Original commit message from CVS:
2005-10-02 Andy Wingo <wingo@pobox.com>
* configure.ac (GST_PLUGIN_LDFLAGS): Change to be like -base.
* ext/flac/gstflacenc.c: Ported to 0.9.
* ext/flac/gstflacdec.c (gst_flacdec_loop): Handle errors better.
* ext/flac/Makefile.am: Add the GST_PLUGINS_BASE cflags and libs,
and link to gsttagedit. Enable flacenc.
* ext/flac/gstflacdec.c: Re-enable tag reading.
Original commit message from CVS:
* gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_setcaps):
* gst/rtp/gstrtpgsmparse.c:
* gst/rtp/gstrtph263penc.c:
* gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_class_init),
(gst_rtpmp4venc_parse_data), (gst_rtpmp4venc_handle_buffer),
(gst_rtpmp4venc_set_property):
* gst/rtp/gstrtpmpaenc.c: (gst_rtpmpaenc_handle_buffer):
Various class and caps fixes from Andre Magalhaes (andrunko)
Original commit message from CVS:
* configure.ac:
Fix unexpanded autoconf macro GST_DOC, which has been renamed
to GST_DOCBOOK_CHECK (see common/m4/gst-doc.m4) (#316202).
Original commit message from CVS:
* sys/oss/gstosssink.c: (gst_oss_sink_prepare):
Fix playback of mono streams (bytes_per_sample should be set
from the sample width and the number of channels negotiated,
and not just be set to 4) (#317338)
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_base_init),
(gst_avi_demux_class_init), (gst_avi_demux_parse_stream),
(gst_avi_demux_change_state):
Don't crash when encountering a stream with an unknown fourcc or
codec id. Instead, create a pad of type video/x-avi-unknown or
audio/x-avi-unknown, which as a side-effect also results in less
confusing error messages in players ('no decoder' vs. 'no streams');
minor fixes to state change function and class_init function.
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_init):
* gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_init):
These are sinks.
Original commit message from CVS:
* check/elements/level.c: (GST_START_TEST):
fix test for new GstClockTime use
* gst/level/gstlevel.c: (gst_level_init), (gst_level_set_caps),
(gst_level_transform_ip):
* gst/level/gstlevel.h:
fix up the decay peak, ensuring the decay peak is never lower
than the peak for that interval
Original commit message from CVS:
* gst/rtp/TODO:
* gst/rtp/gstrtpdec.c: (gst_rtpdec_getcaps):
* gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_class_init),
(gst_rtpmp4venc_parse_data), (gst_rtpmp4venc_handle_buffer),
(gst_rtpmp4venc_set_property):
* gst/rtp/gstrtpmp4venc.h:
* gst/rtp/gstrtpmpaenc.c: (gst_rtpmpaenc_handle_buffer):
* gst/rtp/gstrtpmpaenc.h:
Use is_filled to both check MTU and max-ptime of base class.
Original commit message from CVS:
* gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_class_init),
(gst_rtpmp4venc_parse_data), (gst_rtpmp4venc_handle_buffer),
(gst_rtpmp4venc_set_property):
Don't fragment packets with multiple frames.
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:
* ext/flac/gstflacdec.c: (flac_caps_factory), (raw_caps_factory),
(gst_flacdec_write), (gst_flacdec_convert_src):
* ext/flac/gstflacdec.h:
Add support for flac files with 24/32 bits per sample; and misc.
minor clean-ups. Seeking is still partly broken (for me at least).
Original commit message from CVS:
2005-09-05 Andy Wingo <wingo@pobox.com>
* ext/dv/gstdvdemux.c (gst_dvdemux_chain): Move the pad adding
here from the state change handler, so we fire signals without
holding the state lock.
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:
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:
* 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.
