Original commit message from CVS:
* ext/speex/gstspeexenc.c: (gst_speexenc_get_tag_value),
(comment_init), (comment_add):
Fix handling of GST_TAG_DATE, which is now of GST_TYPE_DATE;
use GST_READ_UINT32_LE() and friends rather than the private
implementation of those same macros.
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.
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.
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:
* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_decode_indirect),
(gst_jpeg_dec_decode_direct), (gst_jpeg_dec_chain):
Fix decoding of pictures with certain uneven or unaligned
widths where jpeglib needs more horizontal padding than our
I420 buffers provide, resulting in blocky artifacts at the
left side of the picture (#164176).
Also make use of our shiny new GST_ROUND_N() macros.
Original commit message from CVS:
* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_init), (gst_jpeg_dec_chain),
(gst_jpeg_dec_change_state):
* ext/jpeg/gstjpegdec.h:
Fix crashes/invalid memory access for pictures that have a height
that is not a multiple of 16 (or rather: v_samp_factor * DCTSIZE).
Also fix the state change function for downwards state changes
(need to chain up to parent before destroying our resources, to
make sure pads get deactivated and our chain function isn't
running and using those very same resources in another thread).
The jpeg line buffer only needs to be v_samp_factor*DCTSIZE lines
per plane, not picture_height lines; allocate that on the stack.
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:
* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_init),
(gst_jpeg_dec_setcaps), (gst_jpeg_dec_chain),
(gst_jpeg_dec_change_state):
* ext/jpeg/gstjpegdec.h:
Make mjpeg actually work and skip jpeg data parsing if we
know that the input is packetized (ie. each input buffer
is exactly one jpeg frame).
Original commit message from CVS:
* ext/mad/gstmad.c: (gst_mad_init), (gst_mad_chain):
It'd be nice if I could listen to my mp3 files, so send out an
initial discont, as the sink apparently wants.
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:
* ext/faad/gstfaad.c: (gst_faad_base_init), (gst_faad_class_init),
(gst_faad_init), (gst_faad_setcaps), (gst_faad_srcgetcaps),
(gst_faad_event), (gst_faad_update_caps), (gst_faad_chain),
(gst_faad_change_state):
* ext/faad/gstfaad.h:
Fix negotiation (#310932) and miscellaneous other stuff. Probably
still needs some more work.
Original commit message from CVS:
* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_init),
(gst_jpeg_dec_setcaps), (gst_jpeg_dec_chain):
Add setcaps() function (for mjpeg).
Original commit message from CVS:
2005-08-08 Andy Wingo <wingo@pobox.com>
* ext/esd/esdsink.c (gst_esdsink_getcaps): Seems that wierd
va_list caps setting function was borked. Fixed esdsink.
Original commit message from CVS:
2005-08-08 Andy Wingo <wingo@pobox.com>
* sys/oss/gstosssink.c (gst_oss_sink_open, gst_oss_sink_close)
(gst_oss_sink_prepare, gst_oss_sink_unprepare): Update for newer
audiosink api.
* ext/raw1394/gstdv1394src.c (gst_dv1394src_get_property)
(gst_dv1394src_set_property): Style. All about the style.
* ext/esd/esdsink.c (gst_esdsink_getcaps): Return specific caps
only if in READY or higher (i.e., if _open() has been called.)
(gst_esdsink_open, gst_esdsink_close, gst_esdsink_prepare)
(gst_esdsink_unprepare): Update for audiosink changes.
(gst_esdsink_change_state): Die!
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:
* ext/Makefile.am:
* ext/libpng/Makefile.am:
* ext/libpng/gstpng.c:
* ext/libpng/gstpngenc.c:
Ported pngenc , still have to port pngdec...
Original commit message from CVS:
* ext/mad/gstmad.c: (gst_mad_src_event):
First try forwarding events, makes seeking in AVI files with mp3
audio work again.
Original commit message from CVS:
2005-07-19 Edgard N. A. G. Lima <edgard.lima@indt.org.br>
* configure.ac
* ext/Makefile.am
* ext/amrnb/amrnbdec.c
* ext/amrnb/amrnbenc.c
* ext/amrnb/amrnbparse.c
* ext/faad/gstfaad.c
* ext/mpeg2dec/gstmpeg2dec.c
Ported amrnb, faad, mpeg2dec to 0.9
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:
2005-07-19 Andy Wingo <wingo@pobox.com>
* ext/dv/gstdvdec.c (gst_dvdec_decode_video): Set the proper
framerate on the outbound buffer.
Original commit message from CVS:
2005-07-19 Andy Wingo <wingo@pobox.com>
* ext/dv/gstdvdec.c (gst_dvdec_decode_video): Don't clobber
alloc_buffer's return value.
(gst_dvdec_decode_frame): Handle unlinked pads with grace and
agility.
Original commit message from CVS:
* gst/realmedia/rmdemux.c: (gst_rmdemux_sink_event),
(gst_rmdemux_change_state), (gst_rmdemux_chain),
(gst_rmdemux_get_stream_by_id), (gst_rmdemux_send_event),
(gst_rmdemux_add_stream):
Send discont event before pushing first buffer.
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:
2005-07-04 Andy Wingo <wingo@pobox.com>
* ext/aalib/gstaasink.c (gst_aasink_fixate): Update for newer
fixate prototype.
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>
* configure.ac:
* ext/mad/Makefile.am:
* sys/oss/Makefile.am:
Roll gstreamer-interfaces-0.9.pc into gstreamer-plugins-base-0.9.pc
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:
2005-06-29 Andy Wingo <wingo@pobox.com>
* gst/videofilter/gstvideoexample.c: Removed gst_library_load, I
think. Whatever this plugin actually does, that I don't know.
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.
