Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (group_commit), (new_decoded_pad),
(setup_substreams), (set_active_source):
Unlock GROUP_LOCK in failure cases, so that we don't deadlock when
trying to go to NULL if we failed to read a file.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiotestsrc_class_init), (gst_audiotestsrc_get_times),
(gst_audiotestsrc_create):
* gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
(gst_sinesrc_get_times), (gst_sinesrc_create):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_videotestsrc_class_init), (gst_videotestsrc_get_times),
(gst_videotestsrc_create):
The base class can now sync for us.
Original commit message from CVS:
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate):
Check if the caps have a fourcc field. Fixes crash for
gst-launch-0.9 v4lsrc name=source autoprobe=false autoprobe-fps=false copy-mode=1 device=/dev/video0 ! ffmpegcolorspace ! "video/x-raw-yuv, format=(fourcc)I420" ! xvimagesink
Original commit message from CVS:
* gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
(gst_sinesrc_get_query_types), (gst_sinesrc_src_query),
(gst_sinesrc_newsegment):
Send newsegment event in TIME format, set duration if
num-buffers is set, fix duration querying.
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (gst_ogg_mux_request_new_pad),
(gst_ogg_mux_push_buffer), (gst_ogg_mux_dequeue_page),
(gst_ogg_mux_pad_queue_page), (gst_ogg_mux_queue_pads),
(gst_ogg_mux_collected):
Fix EOS handling, partially. Now forwarding an EOS event once we have
EOS on all pads works correctly. However, we still don't properly set
EOS on the actual ogg stream pages.
Original commit message from CVS:
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_init),
(gst_base_rtp_depayload_set_gst_timestamp):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
We need to send a newsegment event for each instance, not
just for the first instance of this class (get rid of
static variable in function). (#321011).
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (gst_ogg_mux_request_new_pad),
(gst_ogg_mux_buffer_from_page), (gst_ogg_mux_push_buffer),
(gst_ogg_mux_dequeue_page), (gst_ogg_mux_pad_queue_page),
(gst_ogg_mux_send_headers), (gst_ogg_mux_collected):
Forward port rewrite of muxing strategy to 0.9 version of oggmux.
This makes us mux things correctly according to the ogg muxing
rules. Still not handling EOS correctly right now, though.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
(gst_ogg_pad_submit_packet), (gst_ogg_chain_new):
Initialise segment_stop to GST_CLOCK_TIME_NONE when
creating a new chain; should fix live streaming. Also
add more debug output and fix a typo.
Original commit message from CVS:
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
* gst/volume/gstvolume.c: (volume_set_caps):
Fix compilation on Solaris with Forte. (#320923)
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
(gst_decode_bin_dispose), (free_dynamics), (remove_fakesink),
(pad_blocked), (close_pad_link), (new_pad), (no_more_pads):
Handle the case where a pad_block failed.
Original commit message from CVS:
2005-10-31 Michael Smith <msmith@fluendo.com>
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_init),
(gst_ogg_demux_read_chain), (gst_ogg_demux_read_end_chain),
(gst_ogg_demux_collect_chain_info), (gst_ogg_print):
Patch from Alessandro Decina <alessandro@nnva.org>.
Make oggdemux only find the final time in a chain, not per-pad,
since the per-pad information can be very expensive to locate, and
it isn't used anywhere. This makes reading a file containing
OggSkeleton reasonably fast.
Also, make chain finding work when there are logical bitstreams that
can't be decoded. Fixes#319110.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.h:
Don't break ABI.
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_caps_to_pixfmt):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_set_caps):
Some more comments.
Handle missing required caps fields better.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_get_offset),
(gst_base_audio_sink_render):
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
(gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
(gst_ring_buffer_pause_unlocked), (gst_ring_buffer_pause),
(gst_ring_buffer_stop), (wait_segment), (gst_ring_buffer_commit),
(gst_ring_buffer_read):
* gst-libs/gst/audio/gstringbuffer.h:
Add flushing mode to the ringbuffer so that it in all cases does
not try to handle more audio. This makes sure it does not try to
block anymore when flushing and fixes a livelock.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_query_convert),
(gst_ogg_demux_chain_peer), (gst_ogg_demux_perform_seek),
(gst_ogg_demux_read_chain), (gst_ogg_demux_read_end_chain):
Explicitly check for -1 values before doing a conversion
and always map them to -1. (#315545)
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query), (gst_adder_collected),
(gst_adder_change_state):
Fix timestamps and fix deadlock when stopping the collectpads.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (audio_convert_clean_context):
When clearing an audioconvert context, set tmpbufsize to zero, so
we'll allocate it again later if required.
This fixes audioconvert re-negotiating formats, which previously
segfaulted with a NULL destination buffer.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
Correctly flush decoder samples even if we could not
copy them to an output buffer. Fixes#319618.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_offset), (gst_base_audio_sink_render):
Remove g_print
Use sync property from baseclass to disable sync.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_offset), (gst_base_audio_sink_render):
Buffers with no timestamps get aligned with previous buffers or
on underrun, played ASAP.
Original commit message from CVS:
2005-10-24 Julien MOUTTE <julien@moutte.net>
* gst-libs/gst/video/video.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
And
here comes my change on caps for framerate and geometry range.
We are now accepting 1 to MAXINT for width and height, and from
0.0 to MAXDOUBLE for framerate. That allows duration less png
frames
to be blended correctly in videomixer.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
(gst_decode_bin_dispose), (free_dynamics), (pad_unblocked),
(pad_blocked), (close_pad_link), (new_pad):
Don't try to remove elements twice.
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_dec_src_query),
(theora_dec_sink_event):
* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
(vorbis_dec_sink_event), (vorbis_handle_identification_packet),
(vorbis_handle_data_packet):
* ext/vorbis/vorbisdec.h:
Fix old naming.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Don't try to sync on buffers without a timestamp.
