Original commit message from CVS:
* ext/esd/esdsink.c: (gst_esdsink_class_init),
(gst_esdsink_finalize):
Use a finalize function, not dispose, and more importantly,
call the parent class finalize function too
Original commit message from CVS:
* ext/dv/gstdvdec.c: (gst_dvdec_handle_sink_event):
Set EOS on the element when processing an EOS event.
* ext/speex/gstspeexdec.h:
* ext/speex/gstspeexenc.h:
Only keep a const ptr to the mode
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_caps_with_data),
(gst_riff_create_audio_template_caps):
Allow WMAV3, with up to 6 channels.
* gst/asfdemux/gstasfmux.c: (gst_asfmux_request_new_pad):
Don't call gst_pad_set_event_function on a sink pad.
* gst/mpegstream/gstdvddemux.c:
(gst_dvd_demux_get_subpicture_stream),
(gst_dvd_demux_set_cur_audio), (gst_dvd_demux_set_cur_subpicture):
Copy the explicit caps that were set across to the cur_* pads,
instead of trying to use a possibly non-existent negotiated caps.
Reset the type of subpicture pads to UNKNOWN after calling init_stream,
so that the caps get set.
Original commit message from CVS:
* ext/dv/gstdvdec.c: (gst_dvdec_video_getcaps),
(gst_dvdec_video_link), (gst_dvdec_push), (gst_dvdec_loop):
Allow a little margin when negotiating the framerate.
Original commit message from CVS:
* ext/speex/gstspeexdec.c: (gst_speex_dec_init),
(speex_dec_convert):
sinkconvert function so oggdemux can get the file length (totem).
Original commit message from CVS:
* ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_video_getcaps),
(gst_dvdec_video_link), (gst_dvdec_push), (gst_dvdec_loop):
* ext/dv/gstdvdec.h:
Make sure we renegotiate aspect ratio when the camera switches.
Original commit message from CVS:
* ext/dv/gstdvdec.c: (gst_dvdec_video_getcaps),
(gst_dvdec_video_link), (gst_dvdec_push):
* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_class_init),
(gst_smokeenc_resync), (gst_smokeenc_chain):
Fix mimetype on smoke encoder.
Add aspect ratio to dvdec. Not sure if these
values are correct though....
Original commit message from CVS:
* ext/speex/gstspeexenc.c: (gst_speexenc_class_init),
(gst_speexenc_chain):
Fix speex timestamps so that it gets muxed properly.
Original commit message from CVS:
* ext/raw1394/gstdv1394src.c: (gst_dv1394src_get_type),
(gst_dv1394src_base_init), (gst_dv1394src_class_init),
(gst_dv1394src_init), (gst_dv1394src_dispose),
(gst_dv1394src_iso_receive), (gst_dv1394src_discover_avc_node),
(gst_dv1394src_change_state), (gst_dv1394src_get_event_mask),
(gst_dv1394src_event), (gst_dv1394src_get_formats),
(gst_dv1394src_convert), (gst_dv1394src_get_query_types),
(gst_dv1394src_query), (gst_dv1394src_uri_get_type),
(gst_dv1394src_uri_get_protocols), (gst_dv1394src_uri_get_uri),
(gst_dv1394src_uri_set_uri), (gst_dv1394src_uri_handler_init):
* ext/raw1394/gstdv1394src.h:
Added conversion/query functions.
Update buffer timestamps,
Added signals.
Added uri dv:// so it might play from the firewire in playbin.
Fix a possible leak.
Added debugging.
Original commit message from CVS:
* ext/raw1394/gstdv1394src.c: (gst_dv1394src_class_init),
(gst_dv1394src_init), (gst_dv1394src_set_property),
(gst_dv1394src_get_property), (gst_dv1394src_iso_receive),
(gst_dv1394src_discover_avc_node), (gst_dv1394src_change_state):
* ext/raw1394/gstdv1394src.h:
Added AV/C VTR control support needed for some cameras.
Added automatic port detection.
Added properties for selecting the channel.
The configure.ac script is not yet updated to reflect the
new libavc1394 and librom1394 dependencies.
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flacdec_src_query):
Only return true if we actually filled something in. Prevents
player applications from showing a random length for flac files.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init),
(gst_riff_read_use_event), (gst_riff_read_handle_event),
(gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh),
(gst_riff_read_strf_vids_with_data),
(gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs):
OK, ok, so I implemented event handling. Apparently it's normal
that we receive random events at random points without asking
for it.
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
(gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index),
(gst_avi_demux_sync), (gst_avi_demux_stream_scan),
(gst_avi_demux_massage_index), (gst_avi_demux_stream_header),
(gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry),
(gst_avi_demux_stream_data), (gst_avi_demux_loop):
* gst/avi/gstavidemux.h:
Implement non-lineair chunk handling and subchunk processing.
The first solves playback of AVI files where the audio and video
data of individual buffers that we read are not synchronized.
This should not happen according to the wonderful AVI specs, but
of course it does happen in reality. It is also a prerequisite for
the second. Subchunk processing allows us to cut chunks in small
pieces and process each of these pieces separately. This is
required because I've seen several AVI files with incredibly large
audio chunks, even some files with only one audio chunk for the
whole file. This allows for proper playback including seeking.
This patch is supposed to fix all AVI A/V sync issues.
* gst/flx/gstflxdec.c: (gst_flxdec_class_init),
(flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop):
Work.
* gst/modplug/gstmodplug.cc:
Proper return value setting for the query() function.
* gst/playback/gstplaybasebin.c: (setup_source):
Being in non-playing state (after, e.g., EOS) is not necessarily
a bad thing. Allow for that. This fixes playback of short files.
They don't actually playback fully now, because the clock already
runs. This means that small files (<500kB) with a small length
(<2sec) will still not or barely play. Other files, such as mod
or flx, will work correctly, however.
Original commit message from CVS:
* ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_getcaps):
Correct caps negotiation
* gst/volume/gstvolume.c: (volume_chain_float),
(volume_chain_int16):
Modify debug output to be little more informative
* sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
(gst_xvimagesink_xvimage_destroy):
Add XSync calls after detaching from the shared memory segment to
avoid a crash.
Original commit message from CVS:
* ext/jpeg/gstjpegdec.c (gst_jpegdec_chain): Allocate the buffer
after setting caps. Fixes mysterious segfault. Blessed by Wim.
Original commit message from CVS:
* Trying to correct the GST_CHECK_CONFIGPROG macro. It works perfectly for
I hope i dont break anything.
* Shifting the MAS plugin back from gst-rotten.
Original commit message from CVS:
2004-07-30 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* ext/libpng/gstpngenc.c: (gst_pngenc_class_init):
fix default for newmedia flag
Original commit message from CVS:
2004-07-30 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* ext/libpng/gstpngenc.c: (gst_pngenc_class_init),
(gst_pngenc_init), (gst_pngenc_chain), (gst_pngenc_get_property),
(gst_pngenc_set_property):
* ext/libpng/gstpngenc.h:
Added newmedia support to pngenc so now gst-launch-0.8 videotestsrc ! ffmpegcolorspace ! pngenc snapshot=false newmedia=true ! multifilesink location=blah%d.png works as expected
Original commit message from CVS:
2004-07-28 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* ext/lame/gstlame.c: (gst_lame_chain): send tag events downstream
* ext/shout2/gstshout2.c: (gst_shout2send_protocol_get_type),
(gst_shout2send_get_type), (gst_shout2send_set_clock),
(gst_shout2send_class_init), (gst_shout2send_init),
(set_shout_metadata), (gst_shout2send_set_metadata),
(gst_shout2send_chain), (gst_shout2send_set_property),
(gst_shout2send_get_property), (gst_shout2send_connect),
(gst_shout2send_change_state):
* ext/shout2/gstshout2.h:
- fix for sending mp3 audio to icecast2 server, if pad link function not
called before PAUSED state
- added option to use GStreamer clock sync (as opposed to libshout's own sync)
- added tagging support for mp3 audio broadcasted
* gst/monoscope/gstmonoscope.c: (gst_monoscope_class_init):
debug info
