Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_add),
(gst_multifdsink_remove), (gst_multifdsink_remove_client_link),
(is_sync_frame), (gst_multifdsink_new_client),
(gst_multifdsink_handle_client_write),
(gst_multifdsink_recover_client), (gst_multifdsink_queue_buffer),
(gst_multifdsink_handle_clients), (gst_multifdsink_change_state):
Turn warnings into info.
Don't allow a state change in the streaming thread.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_video_element),
(gen_vis_element), (remove_sinks), (setup_sinks):
Added vis plugin support, need to configure the vis
element to activate it.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (setup_source):
Cleanup the previous pipeline a little earlier for the
case that a source element provides raw data.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (setup_source),
(gst_play_base_bin_change_state):
Actually clean up streaminfo if output fails. This would trigger
if, for example, there was no CD in the drive. No preroll, so
a streaminfo structure is created, but the subsequent state change
of the thread fails.
* gst/playback/gstplaybin.c: (gst_play_bin_change_state):
Don't change state if parent failed.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(gst_play_bin_init), (gst_play_bin_get_property), (handoff),
(gen_video_element), (remove_sinks):
Add small bits of code for screenshot handling.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gst_play_bin_set_property),
(gen_video_element), (gen_audio_element), (setup_sinks):
Don't assume the user provided sinks are named "sink"...
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gen_preroll_element),
(unknown_type), (setup_source), (gst_play_base_bin_remove_element),
(gst_play_base_bin_link_stream):
Do not try to autoplug sources that generate raw streams like
cdparanoia.
disconnect the preroll overrun signal when we don't need it anymore.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (play_base_bin_mute_pad),
(gst_play_base_bin_mute_stream), (gst_play_base_bin_link_stream):
* gst/playback/gstplaybin.c: (setup_sinks):
Implement muting/unmuting of streams, mute streams that are not
used.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
(find_compatibles), (close_pad_link), (try_to_link_1), (new_pad),
(no_more_pads), (close_link), (type_found):
* gst/playback/gstplaybasebin.c: (new_decoded_pad):
* gst/playback/gstplaybin.c: (gen_video_element):
Do not signal the no_more_pads after the first pad when
we are plugging a non dynamic element with multiple
output pads (like swfdec, dvdec, ...).
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
(find_compatibles), (close_pad_link), (try_to_link_1),
(no_more_pads), (close_link), (type_found):
* gst/playback/gstplaybasebin.c: (new_decoded_pad):
* gst/playback/gstplaybin.c: (gen_video_element),
(gen_audio_element):
Set state on newly added element to READY so that negotiation
can happen ASAP.
Addes some more debug info.
Do not try to plug pads with multiple caps structures or ANY
because it is too dangerous since we do not do dynamic
replugging.
Original commit message from CVS:
* gst/playback/README:
* gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
(gst_decode_bin_init), (find_compatibles), (close_pad_link),
(try_to_link_1), (no_more_pads), (close_link), (type_found):
Add some debug info to decodebin, update README
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:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_add),
(gst_multifdsink_remove), (gst_multifdsink_remove_client_link),
(is_sync_frame), (gst_multifdsink_client_queue_buffer),
(gst_multifdsink_new_client),
(gst_multifdsink_handle_client_write),
(gst_multifdsink_recover_client), (gst_multifdsink_queue_buffer),
(gst_multifdsink_handle_clients):
* gst/tcp/gstmultifdsink.h:
Make syncing to keyframes actually work for new clients and lagging
clients.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (new_decoded_pad):
Only signal the no_more_pads signal when we have
added the stream to our list.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (remove_prerolls),
(new_decoded_pad):
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (setup_sinks):
Don't try to preroll or decode more than one audio/video
track.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_change_state):
Throw error if we failed to find a suitable output. This should
throw an error if we successfully set up a pipeline (e.g. because
we recognized a media file) but found no decodable streams in it
(e.g. because it contains only media stream types for which we
have no decoders, or because it's not a media type).
Original commit message from CVS:
* ext/dirac/Makefile.am:
* ext/dirac/gstdirac.cc:
* ext/dirac/gstdiracdec.cc:
* ext/dirac/gstdiracdec.h:
Do something. Don't actually know if this works because I don't
have a demuxer yet.
* ext/gsm/gstgsmdec.c: (gst_gsmdec_getcaps):
Add channels=1 to caps returned from _getcaps().
* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_get_type),
(gst_ogm_video_parse_get_type), (gst_ogm_audio_parse_base_init),
(gst_ogm_video_parse_base_init), (gst_ogm_parse_init),
(gst_ogm_audio_parse_init), (gst_ogm_video_parse_init),
(gst_ogm_parse_sink_convert), (gst_ogm_parse_chain),
(gst_ogm_parse_change_state):
Separate between audio/video so ogmaudioparse actually uses the
audio pad templates. Both audio and video work now, including
autoplugging. Also use sometimes-srcpad hack.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek):
Handle events better. Don't hang on infinite loops.
* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
(gst_avi_demux_init), (gst_avi_demux_reset),
(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
(gst_avi_demux_stream_header), (gst_avi_demux_stream_data),
(gst_avi_demux_change_state):
* gst/avi/gstavidemux.h:
Improve A/V sync. Still not perfect.
* gst/matroska/ebml-read.c: (gst_ebml_read_seek),
(gst_ebml_read_skip):
Handle events better.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event),
(gst_qtdemux_loop_header), (qtdemux_parse_trak),
(qtdemux_audio_caps):
Add IMA4. Improve event handling. Save offset after a seek when
the headers are at the end of the file so that we don't end up in
an infinite loop.
* gst/typefind/gsttypefindfunctions.c: (qt_type_find):
Add low-priority typefind support for files with no length.
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:
* gst/asfdemux/gstasfdemux.c: (_read_var_length), (_read_guid),
(gst_asf_demux_process_segment), (gst_asf_demux_handle_data),
(gst_asf_demux_process_chunk), (gst_asf_demux_handle_sink_event):
Prevent infinite loops. More correct error reporting.
* gst/auparse/gstauparse.c: (gst_auparse_chain):
Error out if negotiation fails.
* gst/playback/gstplaybasebin.c: (setup_source),
(gst_play_base_bin_change_state), (gst_play_base_bin_error),
(gst_play_base_bin_found_tag):
Error/tag forwarding. Pre-roll fixes for source errors on state
changes (e.g. "file does not exist") to prevent hangs.
Original commit message from CVS:
* ext/mad/gstmad.c: (gst_mad_check_caps_reset),
(gst_mad_change_state):
Allow for mp3 rate/channels changes. However, only very
conservatively. Reason that we *have* to enable this is smiply
because the mad find_sync() function is not good enough, it will
regularly sync on random data as valid frames and therefore make
us provide random caps as *final* caps of the stream. The best fix
I could think of is to simply require several of the same stream
changes in a row before we change caps.
The actual testcase that works now is #
* ext/ogg/Makefile.am:
* ext/ogg/gstogg.c: (plugin_init):
* ext/ogg/gstogmparse.c:
OGM support (video only for now; I need an audio sample file).
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_base_init),
(gst_asf_demux_process_stream), (gst_asf_demux_video_caps),
(gst_asf_demux_add_video_stream):
WMV extradata.
* gst/playback/gstplaybasebin.c: (unknown_type):
Don't error out on single unknown-types after all. It's wrong.
If we found type of video and audio but not of a subtitle stream,
it will still error out (which is unwanted). Will find a better fix
later on.
* gst/typefind/gsttypefindfunctions.c: (ogmvideo_type_find),
(ogmaudio_type_find), (plugin_init):
OGM support.
Original commit message from CVS:
* gst/tcp/gstfdset.c: (gst_fdset_fd_has_closed),
(gst_fdset_fd_has_error), (gst_fdset_fd_can_read),
(gst_fdset_fd_can_write), (gst_fdset_wait):
* gst/tcp/gstmultifdsink.c: (gst_client_status_get_type),
(gst_multifdsink_init), (gst_multifdsink_add),
(gst_multifdsink_remove), (gst_multifdsink_get_stats),
(gst_multifdsink_remove_client_link),
(gst_multifdsink_client_queue_buffer),
(gst_multifdsink_handle_client_write),
(gst_multifdsink_recover_client), (gst_multifdsink_handle_clients),
(gst_multifdsink_close), (gst_multifdsink_change_state):
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcpserversink.c: (gst_tcpserversink_class_init),
(gst_tcpserversink_removed):
Small cleanups in fdset.c
Use a hastable to map fd to the client structure for faster
lookup in _remove and get_stats.
Added virtual function to close the fds.
Handle clients even when the select/poll call was unblocked because
of a command.
Implement syncing to keyframe in the recovery procedure.
Original commit message from CVS:
* configure.ac: remove NASM check, since we don't use it. Update
dirac check to 0.4
* ext/dirac/gstdiracdec.cc: update to current 0.4 API
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
Initialized variables.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_change_state),
(gst_qtdemux_loop_header), (qtdemux_parse), (qtdemux_parse_trak),
(gst_qtdemux_handle_esds), (qtdemux_audio_caps): Fix seeking, add
SVQ3 format
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_remove_client_link),
(gst_multifdsink_handle_clients), (gst_multifdsink_change_state):
Don't close the fd in multifdsink as we didn't open it in the
first place. Some cleanups.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (state_change), (setup_source),
(gst_play_base_bin_change_state):
Handle the case where we failed to setup a clear pipeline. This
will throw an error (or EOS, another nice case) and if you don't
catch that, the app will wait for the signal forever (and thus
hang).
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssink.c:
(gst_gnomevfssink_uri_get_protocols):
* ext/gnomevfs/gstgnomevfssrc.c:
(gst_gnomevfssrc_uri_get_protocols):
* ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
* ext/gnomevfs/gstgnomevfsuri.h:
Use _uri_new() instead of _open(), so it doesn't take as long and
Christophe's computer won't hang.
* gst/playback/gstplaybasebin.c: (unknown_type):
Throw error on unknown media type, so apps actually display it.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (queue_overrun), (no_more_pads),
(setup_source), (gst_play_base_bin_set_property),
(gst_play_base_bin_add_element):
* gst/playback/gstplaybin.c: (gst_play_bin_send_event):
Some more work on making sure seeking pauses the pipeline and
that changing the uri actually does something.
Original commit message from CVS:
* gst/tcp/gstfdset.c: (gst_fdset_wait):
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_close):
* gst/tcp/gsttcpserversink.c: (gst_tcpserversink_init_send),
(gst_tcpserversink_close):
Be a bit more paranoid when freeing memory.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
(gst_play_base_bin_dispose), (gst_play_base_bin_set_property):
Handle double disposals, and proper change of URIs.
Original commit message from CVS:
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_update),
(gst_alsa_mixer_get_volume), (gst_alsa_mixer_set_volume),
(gst_alsa_mixer_set_mute), (gst_alsa_mixer_set_record),
(gst_alsa_mixer_set_option), (gst_alsa_mixer_get_option):
Update mixer (to sync with other sessions) if we try to obtain
a new value. This makes alsamixer work accross applications.
* ext/alsa/gstalsasink.c: (gst_alsa_sink_get_time):
Only call sync functions if we're running, else alsalib asserts.
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_src_query):
Sometimes fails to compile. Possibly a gcc bug.
* gst/playback/gstplaybin.c: (gen_video_element),
(gen_audio_element):
Add a reference to an application-provided object, because we lose
this same reference if we add it to the bin. If we don't do this,
we can only use this object once and thus crash if we go from
ready to playing, back to ready and back to playing again.
Also add an audioscale element because several cheap soundcards -
like mine - don't support all samplerates.
* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get),
(gst_ximagesink_xcontext_clear), (gst_ximagesink_change_state):
Fix wrong order or PAR calls. Makes automatically obtained PAR
from the X server atually being used.
Original commit message from CVS:
* gst/tcp/gstfdset.c: (gst_fdset_free), (gst_fdset_set_mode),
(gst_fdset_get_mode), (gst_fdset_add_fd), (gst_fdset_remove_fd),
(gst_fdset_fd_ctl_write), (gst_fdset_fd_ctl_read),
(gst_fdset_fd_has_closed), (gst_fdset_fd_has_error),
(gst_fdset_fd_can_read), (gst_fdset_fd_can_write),
(gst_fdset_wait):
* gst/tcp/gstfdset.h:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_add),
(gst_multifdsink_client_queue_buffer),
(gst_multifdsink_handle_client_write):
* gst/tcp/gstmultifdsink.h:
Some extra checks in gstfdset.
Only use send() when the fd is a socket. Don't try to
read from write only fds.
Original commit message from CVS:
* gst/tcp/gstfdset.c: (ensure_size), (gst_fdset_wait):
Realloc test fdset in the lock and right before starting
the poll call. Bump the limit to 4096.
Original commit message from CVS:
2004-08-17 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* gst/audioscale/gstaudioscale.c:
* gst/audioscale/gstaudioscale.h:
made audioscale resample from any sample rate to any sample rate
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
(gst_multifdsink_init), (gst_multifdsink_add),
(gst_multifdsink_client_queue_buffer),
(gst_multifdsink_set_property), (gst_multifdsink_get_property):
* gst/tcp/gstmultifdsink.h:
Added option to send a keyframe to clients as the first buffer.
Make timeout property writable.
Original commit message from CVS:
* gst/tcp/gstfdset.c: (ensure_size), (gst_fdset_new),
(gst_fdset_add_fd), (gst_fdset_remove_fd),
(gst_fdset_fd_has_closed), (gst_fdset_fd_has_error),
(gst_fdset_fd_can_read), (gst_fdset_fd_can_write),
(gst_fdset_wait):
Make sure the pollfds are not changed when the poll call is
running. Protect against array out of bounds.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_unit_type_get_type),
(gst_client_status_get_type), (gst_multifdsink_class_init),
(gst_multifdsink_init), (gst_multifdsink_remove_client_link),
(gst_multifdsink_handle_client_read),
(gst_multifdsink_handle_client_write),
(gst_multifdsink_recover_client), (gst_multifdsink_queue_buffer),
(gst_multifdsink_handle_clients), (gst_multifdsink_set_property),
(gst_multifdsink_get_property):
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcp-marshal.list:
Starting to prepare for specifying buffer time in other units
than buffers. Expose remove reason in signal.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_add),
(gst_multifdsink_remove), (gst_multifdsink_clear),
(gst_multifdsink_remove_client_link),
(gst_multifdsink_handle_client_read),
(gst_multifdsink_client_queue_data),
(gst_multifdsink_client_queue_buffer),
(gst_multifdsink_handle_client_write),
(gst_multifdsink_queue_buffer), (gst_multifdsink_handle_clients),
(gst_multifdsink_chain), (gst_multifdsink_close):
* gst/tcp/gstmultifdsink.h:
Added more debugging info. Changed the way clients are
removed from the lists. Fixed a bug where a bad file descriptor
could cause many clients to be removed.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
(gst_multifdsink_add), (gst_multifdsink_get_stats),
(gst_multifdsink_client_remove),
(gst_multifdsink_handle_client_read),
(gst_multifdsink_handle_client_write),
(gst_multifdsink_queue_buffer), (gst_multifdsink_handle_clients):
Do a bit more logging, make the client_read code more robust.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
(gst_multifdsink_add), (gst_multifdsink_get_stats),
(gst_multifdsink_client_remove),
(gst_multifdsink_handle_client_read),
(gst_multifdsink_handle_client_write),
(gst_multifdsink_queue_buffer), (gst_multifdsink_handle_clients):
Make sure we don't try to read more from a client that what
ioctl says us or we deadlock.
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c:
(gst_videotestsrc_get_capslist), (generate_capslist),
(plugin_init):
generate the list of supported caps at startup and reuse it instead
of always generating it
Original commit message from CVS:
* gst/audioscale/gstaudioscale.c:
- fix templates to only support S16, it's the only format that works
- make caps nego code use try_set_caps_nonfixed and fixation instead
of try_set_caps twice, which is not nice for autopluggers
- change rank to secondary, so autopluggers can pick it up after
audioconvert
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
(gst_multifdsink_add), (gst_multifdsink_remove),
(gst_multifdsink_clear), (gst_multifdsink_get_stats),
(gst_multifdsink_client_remove),
(gst_multifdsink_handle_client_write),
(gst_multifdsink_queue_buffer), (gst_multifdsink_handle_clients):
* gst/tcp/gstmultifdsink.h:
Recover from a select with a bad file descriptor by removing
the client.
Original commit message from CVS:
* gst/tcp/gsttcpclientsrc.c (gst_tcpclientsrc_get): Make sure that
the pad is negotiated.
* gst/ffmpegcolorspace/gstffmpegcolorspace.c (gst_ffmpegcolorspace_chain): Ditto
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init): Add typefind
for ELF files, since they can easily be recognized as audio/mpeg.
(bug #147441)
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
(gst_multifdsink_add), (gst_multifdsink_get_stats),
(gst_multifdsink_client_remove),
(gst_multifdsink_handle_client_write),
(gst_multifdsink_queue_buffer):
* gst/tcp/gstmultifdsink.h:
More multifdsink stats. Avoid deadlock by releasing locks
before sending out a signal.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
(gst_multifdsink_init), (gst_multifdsink_add),
(gst_multifdsink_client_remove),
(gst_multifdsink_handle_client_write),
(gst_multifdsink_queue_buffer), (gst_multifdsink_chain),
(gst_multifdsink_set_property), (gst_multifdsink_get_property),
(gst_multifdsink_init_send):
* gst/tcp/gstmultifdsink.h:
Added more stats, added timeout for a client, fixed some typos
and added some comments.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data):
mp42/mp43 (no caps) exist too.
* gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps):
Set pixel_width/height; we've got them in-caps.
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
* gst/wavparse/gstwavparse.c: (plugin_init):
Both are valid primary.
* sys/oss/gstossmixer.c:
Remove i18n hack and enable translations.
Original commit message from CVS:
2004-07-11 Andy Wingo <wingo@pobox.com>
* gst/audioconvert/gstaudioconvert.c (gst_audio_convert_link): For
float, "any" caps -> buffer_frames=[0,MAX].
* gst/interleave/interleave.c (interleave_getcaps): Seems the core
doesn't intersect our caps with the template any more. Do it
ourselves.
(interleave_buffered_loop): Use g_newa instead of malloc/free.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_recover_policy_get_type),
(gst_multifdsink_class_init), (gst_multifdsink_add),
(gst_multifdsink_remove), (gst_multifdsink_clear),
(gst_multifdsink_client_remove),
(gst_multifdsink_handle_client_read),
(gst_multifdsink_client_queue_data),
(gst_multifdsink_client_queue_caps),
(gst_multifdsink_client_queue_buffer),
(gst_multifdsink_handle_client_write),
(gst_multifdsink_recover_client), (gst_multifdsink_queue_buffer),
(gst_multifdsink_handle_clients), (gst_multifdsink_thread),
(gst_multifdsink_init_send), (gst_multifdsink_close):
Fix wrong GList iteration that could crash the server when
more then 2 clients disconnect at the same time. Read all the
pending commands in one batch to recover from command storms under
very heavy load.
Original commit message from CVS:
* gst/tcp/gsttcpserversink.c: (gst_tcpserversink_class_init),
(gst_tcpserversink_init), (gst_tcpserversink_handle_server_read),
(gst_tcpserversink_client_remove),
(gst_tcpserversink_handle_client_read),
(gst_tcpserversink_client_queue_data),
(gst_tcpserversink_client_queue_caps),
(gst_tcpserversink_client_queue_buffer),
(gst_tcpserversink_handle_client_write),
(gst_tcpserversink_queue_buffer),
(gst_tcpserversink_handle_clients), (gst_tcpserversink_thread),
(gst_tcpserversink_chain), (gst_tcpserversink_set_property),
(gst_tcpserversink_get_property), (gst_tcpserversink_init_send),
(gst_tcpserversink_close):
* gst/tcp/gsttcpserversink.h:
Serversink rewrite. Really do non blocking writes to clients and
maintain an internal queue to handle slower clients while not
disturbing fast clients.
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audiorate_link),
(gst_audiorate_init), (gst_audiorate_chain),
(gst_audiorate_set_property), (gst_audiorate_get_property):
* gst/videorate/gstvideorate.c: (gst_videorate_class_init),
(gst_videorate_chain):
Added some logging, fixed an overflow bug in videorate.
Original commit message from CVS:
* gst-libs/gst/colorbalance/Makefile.am:
* gst-libs/gst/mixer/Makefile.am:
* gst-libs/gst/play/Makefile.am:
* gst-libs/gst/tuner/Makefile.am:
* gst/tcp/Makefile.am:
* sys/dxr3/Makefile.am:
don't include -enumtypes.[ch] or -marshal.[ch] files in the disted
tarball.
Also add all *.list files that were missing.
* Makefile.am:
add a distcheck hook to ensure the above doesn't happen again.
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_videorate_class_init),
(gst_videorate_init), (gst_videorate_chain),
(gst_videorate_set_property), (gst_videorate_get_property):
Add property to make videorate silent.
Add property to prefer new frames over old ones.
Original commit message from CVS:
* ext/dvdnav/gst-dvd: Grab the gconf key from the right spot
* gst/debug/gstnavseek.c: (gst_navseek_init),
(gst_navseek_segseek), (gst_navseek_handle_src_event),
(gst_navseek_chain):
* gst/debug/gstnavseek.h: Add 's', 'e' and 'l' keypresses to navseek
to define the start,end and loop parameters of a segment seek.
* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_init),
(gst_videotestsrc_get_event_masks),
(gst_videotestsrc_handle_src_event), (gst_videotestsrc_get):
* gst/videotestsrc/gstvideotestsrc.h:
Add seeking support to videotestsrc
Initialise the timestamp_offset variable.
Original commit message from CVS:
* gst/ffmpegcolorspace/imgconvert.c: (img_convert):
Patch 1.3 broke the ordering of the colorspace info and
made the plugin basically work by coincidence, reodered
the info.
Original commit message from CVS:
2004-06-12 Christophe Fergeau <teuf@gnome.org>
* gst/tags/gstvorbistag.c: replaced a g_warning which I added in my
previous commit with GST_DEBUG
Original commit message from CVS:
2004-06-12 Zaheer Abbas Merali <zaheerabbas@merali.org>
* gst/tcp/gsttcpclientsink.c: (gst_tcpclientsink_init_send):
* gst/tcp/gsttcpclientsink.h:
* gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_init_receive):
* gst/tcp/gsttcpclientsrc.h:
* gst/tcp/gsttcpserversink.c: (gst_tcpserversink_init),
(gst_tcpserversink_handle_server_read),
(gst_tcpserversink_init_send):
* gst/tcp/gsttcpserversink.h:
* gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_init_receive):
* gst/tcp/gsttcpserversrc.h:
Modified the tcp plugins so they are portable (IPv4,IPv6, any future
version of IP)
Original commit message from CVS:
2004-06-10 Christophe Fergeau <teuf@gnome.org>
* gst/tags/gstvorbistag.c: (gst_vorbis_tag_add): make sure parsed
vorbis comments are properly encoded in UTF-8 before adding them
to a GstTagList
Original commit message from CVS:
reviewed by Benjamin Otte <otte@gnome.org>
* gst/adder/gstadder.c: (gst_adder_loop):
properly error out when no negotiation has happened yet. (fixes
#143032)
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c: that's
G_HAVE_GNUC_VARARGS, not G_HAVE_GNU_VARARGS. Should fix compile
problems on several systems.
Original commit message from CVS:
* gst/tcp/gsttcp.c: portability (Solaris 10/FreeBSD)
* gst/tcp/gsttcpclientsrc.h: idem
- define MSG_NOSIGNAL if not done
- include unistd.h for off_t
(fixes#143749)
patch by Andrew Turner <zxombie@hotpop.com>
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_fixate):
fixate nicely even when the peer is not negotiating
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps):
make sure we don't allow depth > width
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_fixate):
fixate endianness to G_BYTE_ORDER as default
* gst/audioscale/gstaudioscale.c:
we don't handle another endianness as host-endianness
Original commit message from CVS:
* ext/vorbis/oggvorbisenc.c: (gst_oggvorbisenc_sinkconnect),
(gst_oggvorbisenc_setup):
properly fail when we can't setup the vorbis encoder due to
unsupported settings
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_sinkconnect),
(gst_vorbisenc_setup):
same
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
fix case where warnings occured when one pad was unlinked while the
other's link function was called
Original commit message from CVS:
* gst/videoscale/videoscale.c: (gst_videoscale_scale_nearest),
(gst_videoscale_scale_nearest_str2),
(gst_videoscale_scale_nearest_str4),
(gst_videoscale_scale_nearest_32bit),
(gst_videoscale_scale_nearest_24bit),
(gst_videoscale_scale_nearest_16bit):
Fix the scaling algorithm and avoid a buffer overflow.
removed the while loop in the scaling function as it
was used for point sampling only.
Original commit message from CVS:
- change sunaudio category to Sink/Audio
- change HAVE_FIONREAD macro to GST_CHECK_FIONREAD
- add conditional include for FIONREAD ioctl on more files
Original commit message from CVS:
* gst/ffmpegcolorspace/imgconvert.c: (img_convert): Fixes for
warnings (bugs, actually) noticed by gcc but not forte.
Original commit message from CVS:
* configure.ac: Add sunaudio
* examples/Makefile.am: make gstplay depend on gconf
* gst/ffmpegcolorspace/gstffmpegcodecmap.c: Remove c99-isms
* gst/ffmpegcolorspace/imgconvert.c: (build_rgb_palette),
(convert_table_lookup), (img_convert): remove c99-isms
* gst/ffmpegcolorspace/imgconvert_template.h: make a constant
unsigned, to fix a warning on Solaris
* gst/mpeg1sys/systems.c: bcopy->memcpy
* gst/rtjpeg/RTjpeg.c: (RTjpeg_yuvrgb8): bcopy->memcpy
* sys/Makefile.am: Add sunaudio
Original commit message from CVS:
* ext/jpeg/gstjpegdec.c: (gst_jpegdec_get_type):
* ext/jpeg/gstjpegenc.c: (gst_jpegenc_get_type),
(gst_jpegenc_getcaps):
move format setting to inner loop
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcolorspace_getcaps):
use GST_PAD_CAPS if available so that we use already negotiated
caps
* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_header),
(qtdemux_parse_moov), (qtdemux_parse):
extra debugging
* sys/qcam/qcam-Linux.c: (qc_lock_wait), (qc_unlock):
* sys/qcam/qcam-os.c: (qc_lock_wait), (qc_unlock):
move hardcoded path to DEFINE
Original commit message from CVS:
* ext/mad/gstid3tag.c: (gst_id3_tag_init):
remove leftover g_print
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
don't try setting only a subset of the caps. We don't want to kill
autoplugging on purpose
Original commit message from CVS:
* gconf/Makefile.am: Fix for non-GNU make
* gst-libs/gst/Makefile.am: Change directory order to handle
GstPlay linking with gstinterfaces
* gst-libs/gst/audio/make_filter: make use of tr portable
* gst-libs/gst/play/Makefile.am: Add intended \
* gst-libs/gst/xwindowlistener/xwindowlistener.c:
(gst_xwin_set_clips): Switch to ISO variadic macro. Use a
function prototype instead of void *.
* gst/ffmpegcolorspace/gstffmpegcodecmap.c: Switch to ISO variadic
macro.
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcolorspace_chain): wrap NULL in GST_ELEMENT_ERROR call
* gst/videofilter/make_filter: make use of tr portable
* pkgconfig/Makefile.am: Remove GNU extension in Makefile target
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssink.c:
(_gst_boolean_allow_overwrite_accumulator),
(gst_gnomevfssink_class_init):
fix erase signal - if any handler returns false the file will not be
overwritten. If no handler is connected, the file will not be
overwritten either.
renamed signal to "allow-overwrite"
* ext/mad/gstid3tag.c: (tag_list_to_id3_tag_foreach):
free string when adding it to ID3 failed
* ext/vorbis/vorbisdec.c: (vorbis_dec_event):
unref event when done
* gst/audioconvert/gstaudioconvert.c: (_fixate_caps_to_int):
free caps
* gst/typefind/gsttypefindfunctions.c:
(mpeg_video_stream_type_find):
fix invalid read
Original commit message from CVS:
2004-04-09 Andy Wingo <wingo@pobox.com>
* gst/audioconvert/bufferframesconvert.c: New element to convert
buffer-frames for float streams. Not working nicely yet.
* gst/audioconvert/plugin.h:
* gst/audioconvert/plugin.c: New files.
* gst/audioconvert/Makefile.am: Build the new files.
* gst/audioconvert/gstaudioconvert.c: Initialize via plugin.[ch].
Original commit message from CVS:
* gst/audioscale/gstaudioscale.c: (gst_audioscale_expand_value),
(gst_audioscale_getcaps): Fix getcaps to expand and union lists.
(bug #138225)
* gst/debug/Makefile.am:
* gst/debug/breakmydata.c: (gst_break_my_data_plugin_init):
* gst/debug/gstdebug.c: (plugin_init): Merge elements into one
plugin.
* gst/debug/negotiation.c: (gst_gst_negotiation_get_type),
(gst_negotiation_base_init), (gst_negotiation_class_init),
(gst_negotiation_init), (gst_negotiation_getcaps),
(gst_negotiation_pad_link), (gst_negotiation_chain),
(gst_negotiation_set_property), (gst_negotiation_get_property),
(gst_negotiation_plugin_init): New element to talk about random
negotiation things happening in a pipeline.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_fixate): Don't fixate fields that
aren't in the caps.
* gst/sine/gstsinesrc.c: change rate caps to [1,MAX]
* gst/videocrop/gstvideocrop.c: (plugin_init): Change rank to NONE.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (theora_type_find):
fix bug where typefinding would claim it's theora whenever less then
7 bytes of data were available
Original commit message from CVS:
2004-03-23 Jeremy Simon <jesimon@libertysurf.fr>
* gst/typefind/gsttypefindfunctions.c: (ape_type_find),
(plugin_init):
Add a monkeysaudio typefind function
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link),
(_fixate_caps_to_int), (gst_audio_convert_fixate):
add a fixation function that pretty much does the right thing (fixes
#137556)
Original commit message from CVS:
* gst/sine/gstsinesrc.c: (gst_sinesrc_get):
return taglist correctly from _get function, don't gst_pad_push it.
(fixes#137042)
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_getcaps),
(gst_audio_convert_link), (gst_audio_convert_change_state),
(gst_audio_convert_buffer_from_default_format):
do conversions from/to float correctly, fix some caps nego errors,
export correct supported caps in template and getcaps, use correct
caps in try_set_caps functions
Original commit message from CVS:
2004-03-06 Christophe Fergeau <teuf@gnome.org>
For some reason, I only committed a ChangeLog entry yesterday and
not the corresponding code...
* ext/mad/gstmad.c: Fix detection of Xing headers
* gst/tags/gstid3tag.c: Changes to support TLEN tags
Original commit message from CVS:
* ext/aalib/gstaasink.c: (gst_aasink_fixate), (gst_aasink_init):
Add fixate function. (bug #131128)
* ext/sdl/sdlvideosink.c: (gst_sdlvideosink_init),
(gst_sdlvideosink_fixate): Add fixate function.
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link):
Fix attempt to print a non-pointer using GST_PTR_FORMAT.
* gst/wavparse/gstwavparse.c: (gst_wavparse_parse_fmt):
Fix missing break that was causing ulaw to be interpreted as
raw int.
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_init), (gst_faad_srcgetcaps),
(gst_faad_chain): Fix negotiation.
* ext/librfb/gstrfbsrc.c: (gst_rfbsrc_handle_src_event): Add
key and button events.
* gst-libs/gst/floatcast/floatcast.h: Fix a minor bug in this
dung heap of code.
* gst-libs/gst/gconf/gstreamer-gconf-uninstalled.pc.in: gstgconf
depends on gconf
* gst-libs/gst/gconf/gstreamer-gconf.pc.in: same
* gst-libs/gst/play/play.c: (gst_play_pipeline_setup),
(gst_play_video_fixate), (gst_play_audio_fixate): Add a fixate
function to encourage better negotiation, particularly between
audioconvert and osssink.
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain):
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak): Make some debugging
more important.
* gst/typefind/gsttypefindfunctions.c: Fix mistake in flash
typefinding.
* gst/vbidec/vbiscreen.c: Add glib header
* pkgconfig/gstreamer-play.pc.in: Depends on gst-interfaces.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_channels):
convert channels correctly. convert correctly to unsigned.
Original commit message from CVS:
2004-03-05 Benjamin Otte <otte@gnome.org>
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_buffer_to_default_format):
make float=>int conversion work correctly even in cornercases.
Original commit message from CVS:
2004-02-25 Andy Wingo <wingo@pobox.com>
* gst/interleave/interleave.c (interleave_buffered_loop): Always
push only when channel->buffer is NULL. Prevents segfaults doing
the state change after a nonlocal exit, like a scheme exception.
* gst/audioconvert/gstaudioconvert.c (gst_audio_convert_getcaps):
Handle the case where the intersected caps is empty.
Original commit message from CVS:
2004-02-22 Benjamin Otte <otte@gnome.org>
reported by: Stefan Kost <kost@imn.htwk-leipzig.de>
* gst/audioconvert/gstaudioconvert.c: (plugin_init):
set rank to PRIMARY
* gst/volume/gstvolume.c: (plugin_init):
set rank to NONE
fixes#134960
2004-02-22 Julio M. Merino Vidal <jmmv@menta.net>
reviewed by Benjamin Otte <otte@gnome.org>
* ext/flac/gstflacenc.c: (gst_flacenc_chain):
escape NULL strings in GST_ELEMENT_ERROR properly (fixes#135116)
Original commit message from CVS:
2004-02-20 Andy Wingo <wingo@pobox.com>
* gst/intfloat/, gst/oneton: Removed, replaced by audioconvert and
interleave respectively.
* gst/interleave/deinterleave.c: New plugin: deinterleave
(replaces on oneton).
* gst/interleave/interleave.c: New plugin: interleave.
* gst/interleave/plugin.h: Support file.
* gst/interleave/plugin.c: Support file.
* configure.ac: Remove intfloat and oneton, add interleave.
* ext/sndfile/gstsf.c: Handle events better.
* gst/audioconvert/gstaudioconvert.c: Change to support int2float
and float2int operation. int2float has scheduling problems as
noted in in2float_chain.
Original commit message from CVS:
* ext/gdk_pixbuf/gstgdkpixbuf.c: (plugin_init): Disable gdk_pixbuf
typefinding, since it seems to be worse than nothing.
* gst/typefind/gsttypefindfunctions.c: (qt_type_find): Add ftyp
atom to recognize .mp4 and .m4a files as video/quicktime.
Original commit message from CVS:
* gst/sine/demo-dparams.c: (quit_live),
(dynparm_log_value_changed), (dynparm_value_changed), (main):
Use double dparams, not float.
* gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
(gst_sinesrc_init): Change sync default to FALSE, since multiple
sync'd elements don't really work correctly.
* gst/volume/gstvolume.c: (volume_class_init), (volume_init),
(volume_update_volume), (volume_get_property): Change dparam
to double.
Original commit message from CVS:
* gst/sine/demo-dparams.c: (dynparm_log_value_changed),
(dynparm_value_changed), (main): Convert from float to double.
* gst/sine/gstsinesrc.c: (gst_sinesrc_init): same.
Original commit message from CVS:
Convert a few inner loops to use liboil. This is currently
optional, and is only enabled if liboil is present (duh!).
* configure.ac: Check for liboil-0.1
* gst/intfloat/Makefile.am:
* gst/intfloat/gstint2float.c: (conv_f32_s16), (scalarmult_f32),
(gst_int2float_chain_gint16):
* gst/videofilter/Makefile.am:
* gst/videofilter/gstvideobalance.c: (gst_videobalance_class_init),
(tablelookup_u8), (gst_videobalance_planar411):
* gst/videotestsrc/Makefile.am:
* gst/videotestsrc/gstvideotestsrc.c: (plugin_init):
* gst/videotestsrc/videotestsrc.c: (splat_u8), (paint_hline_YUY2),
(paint_hline_IYU2), (paint_hline_str4), (paint_hline_str3),
(paint_hline_RGB565), (paint_hline_xRGB1555):
Original commit message from CVS:
2004-02-04 Benjamin Otte <in7y118@public.uni-hamburg.de>
* gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
(gst_sinesrc_dispose):
fix memleak by properly disposing sinesrc
Original commit message from CVS:
2004-02-03 Benjamin Otte <in7y118@public.uni-hamburg.de>
* ext/gdk_pixbuf/gstgdkpixbuf.c: (plugin_init):
* gst/typefind/gsttypefindfunctions.c:
fix memleaks shown by gst-typefind
Original commit message from CVS:
2004-02-03 Benjamin Otte <in7y118@public.uni-hamburg.de>
* gst/qtdemux/qtdemux.c: (gst_qtdemux_add_stream):
set explicit caps before adding the element, so the autopluggers can
plug correctly.
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find),
(mpeg2_sys_type_find), (mpeg1_sys_type_find),
(mpeg_video_type_find), (mpeg_video_stream_type_find),
(dv_type_find):
fix memleaks in typefind functions. gst_type_find_suggest takes a const
argument.
Original commit message from CVS:
2004-02-02 Jeremy Simon <jesimon@libertysurf.fr>
* gst/tags/gstvorbistag.c: (gst_vorbis_tag_add),
(gst_tag_to_vorbis_comments):
Add replaygain support to vorbistag
Original commit message from CVS:
2004-01-31 Benjamin Otte <in7y118@public.uni-hamburg.de>
* ext/theora/theoradec.c: (theora_dec_chain):
make comments work
* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
(vorbis_dec_src_event), (vorbis_dec_chain):
add encoder tag, fix tag reading to be more error tolerant, change
BITRATE to NOMINAL_BITRATE, add debugging, don't unref events after
gst_pad_event_default.
* gst/tags/gstvorbistag.c:
(gst_tag_list_from_vorbiscomment_buffer):
undefine function specific define at end of function
Original commit message from CVS:
2004-01-29 Benjamin Otte <in7y118@public.uni-hamburg.de>
* ext/ogg/gstoggdemux.c:
lots of changes - mainly support for chained bitstreams, seeking,
querying and bugfixes of course
* ext/vorbis/Makefile.am:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisdec.h:
add vorbisdec raw vorbis decoder
* ext/vorbis/vorbis.c: (plugin_init):
register vorbisdec as PRIMARY, vorbisfile as SECONDARY
* gst/intfloat/Makefile.am:
* gst/intfloat/float22int.c:
* gst/intfloat/float22int.h:
* gst/intfloat/gstintfloatconvert.c: (plugin_init):
add float2intnew plugin. It converts multichannel interleaved float to
multichannel interleaved int. The name should probably be changed.
* gst/typefind/gsttypefindfunctions.c: (theora_type_find),
(plugin_init):
add typefinding for raw theora video so oggdemux can detect it.
Original commit message from CVS:
2004-01-28 Julien MOUTTE <julien@moutte.net>
* gst-libs/gst/play/gstplay.c: (gst_play_seek_to_time): seek on video
sink element first.
* gst/videoscale/gstvideoscale.c:
(gst_videoscale_handle_src_event): Fixing src event handler.
Original commit message from CVS:
* gst/ac3parse/gstac3parse.c: update to checklist 5
* gst/adder/gstadder.c: rewrite negotiation. update to checklist 5
* gst/audioconvert/gstaudioconvert.c: update to checklist 5
* gst/audioscale/gstaudioscale.c: same
* gst/auparse/gstauparse.c: same
* gst/avi/gstavidemux.c: same
Original commit message from CVS:
2004-01-26 Julien MOUTTE <julien@moutte.net>
* gst/videoscale/gstvideoscale.c: (gst_videoscale_chain): Dunno how
that one managed to stay there... Fixed.
Original commit message from CVS:
2004-01-26 Benjamin Otte <in7y118@public.uni-hamburg.de>
* gst-libs/gst/audio/audio.h:
remove buffer-frames from audio caps
* gst/audioconvert/gstaudioconvert.c:
fix plugin to really work.
