Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
(gst_alsasink_class_init), (gst_alsasink_init),
(gst_alsasink_write), (gst_alsasink_reset):
* ext/alsa/gstalsasink.h:
Add lock to protect alsa calls.
Implement reset to flush samples ASAP, does not work
with dmix though.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_provide_clock),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_render):
Don't try to provide a clock when we are not negotiated since
we might not be able to make it run.
Original commit message from CVS:
* gst-libs/gst/audio/TODO:
Updated.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_event):
On EOS, wait till the last sample is played before posting EOS.
Original commit message from CVS:
2006-02-01 Philippe Kalaf <burger at speedy dot org>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
Patch by Kai Vehmanen : Adds ability to enable newsegment bypass by
setting queue_delay to zero. Also avoid thread being started if
queue_delay is zero.
Original commit message from CVS:
* gst/playback/test6.c: (new_decoded_pad_cb), (show_error), (main):
Make test work again by connecting fakesinks to each decoded pad,
which makes the pipeline wait until each fakesink has a buffer
queued before going to PAUSED state. At that point we know the
decodebin pads are negotiated.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (au_type_find),
(paris_type_find), (ilbc_type_find), (plugin_init):
Fix typefinding for audio/x-au, audio/x-paris and
audio/iLBC-sh. We cannot use the START_WITH macros
here, because there can only be one typefind factory
with the same name (caps), so the second one would
replace the first one and the first one would never
be called when doing typefinding (see #161712).
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_convert),
(vorbis_handle_header_packet), (vorbis_dec_push),
(vorbis_handle_data_packet):
Use scale_int when we can, add some more scaling.
Check packettype before parsing it.
Original commit message from CVS:
* ext/theora/theoradec.c: (_theora_granule_time),
(theora_dec_src_convert), (theora_dec_sink_convert):
Call right _scale functions.
Use parameter instead of some other random value.
Original commit message from CVS:
* ext/theora/theoradec.c: (_theora_granule_frame),
(_theora_granule_time), (_inc_granulepos),
(theora_dec_src_convert), (theora_dec_sink_convert),
(theora_handle_type_packet), (theora_handle_data_packet),
(theora_dec_chain):
Use higher precision timestamps calculation.
Convert some other conversions to _scale.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_create_sine_table), (plugin_init):
* gst/volume/gstvolume.c: (plugin_init):
initialize gst_controller before using
Original commit message from CVS:
* tests/check/pipelines/theoraenc.c:
* tests/check/pipelines/vorbisenc.c:
Define constant using G_GINT64_CONSTANT to avoid errors when
passing it around - otherwise it gets truncated to 32 bits.
Fixes failing tests.
Original commit message from CVS:
2006-01-31 Andy Wingo <wingo@pobox.com>
* sys/v4l/gstv4lsrc.c (gst_v4lsrc_set_caps): Don't segfault if the
caps being set doesn't have a framerate value. Basically a stopgap
measure.
* ext/ogg/gstoggmux.c (GST_BUFFER_END_TIME): New macro. Not
technically correct enough to put into core though.
(gst_ogg_mux_dequeue_page): Use END_TIME instead of TIMESTAMP +
DURATION. Fixes theoraenc ! oggmux.
* sys/v4l/gstv4lsrc.c (gst_v4lsrc_fixate): Fixate to the nearest
fraction, not double.
Original commit message from CVS:
2006-01-30 Andy Wingo <wingo@pobox.com>
* ext/ogg/gstoggmux.c (gst_ogg_mux_dequeue_page): Compare
timestamp + duration, not just timestamp -- ogg pages should be
ordered by stop time. Necessary fix given the change in vorbis
timestamps.
Original commit message from CVS:
2006-01-30 Andy Wingo <wingo@pobox.com>
* ext/theora/theoraenc.c (theora_enc_sink_setcaps)
(gst_theora_enc_init): Pull the granule shift out of the encoder.
(granulepos_add): New function, handles the messiness of adjusting
granulepos values.
(theora_buffer_from_packet):
(theora_enc_chain):
(theora_enc_sink_event): Use granulepos_add, not +.
* tests/check/pipelines/theoraenc.c
(check_buffer_granulepos_from_starttime): Just check the frame
count, not the actual granulepos -- we can't dictate to the
encoder when it should be placing keyframes.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
SERVICE_NOT_AVAILABLE happens for example when you're trying to
play an http:// stream from a server that's not serving
Original commit message from CVS:
2006-01-30 Andy Wingo <wingo@pobox.com>
* tests/check/pipelines/vorbisenc.c (TIMESTAMP_OFFSET):
* tests/check/pipelines/theoraenc.c (TIMESTAMP_OFFSET): Totally
remove the UINT64_CONSTANT macro, doesn't appear to be needed or
available.
Original commit message from CVS:
2006-01-30 Andy Wingo <wingo@pobox.com>
* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c: Same changes as were done to vorbisenc,
although theoraenc was timestamping correctly. Added handling of
streams that start with nonzero timestamps.
* tests/check/Makefile.am:
* tests/check/pipelines/theoraenc.c: New file, basically does same
tests as vorbisenc.
* tests/check/pipelines/vorbisenc.c: I claim these bugs.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_class_init), (gst_audioringbuffer_release),
(gst_audioringbuffer_pause):
Implement pause that does not wait for completion.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Don't drop buffers when going to PAUSED but perform preroll on
remaining samples now that core base class supports this.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_release),
(gst_ring_buffer_pause_unlocked), (gst_ring_buffer_stop),
(gst_ring_buffer_commit):
Pause should not signal waiters.
Implement return value of _commit correctly.
Original commit message from CVS:
2006-01-30 Andy Wingo <wingo@pobox.com>
* tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.
* ext/vorbis/vorbisenc.c (gst_vorbisenc_buffer_from_packet): Logic
updated to timestamp from the first sample, not the last.
(gst_vorbisenc_buffer_from_header_packet): New function, takes
special care of granulepos and timestamp for header packets.
(gst_vorbisenc_chain): Reflow, fix some leaks, and handle the case
when the first buffer has a nonzero timestamp.
* ext/vorbis/vorbisenc.h (GstVorbisEnc.granulepos_offset)
(GstVorbisEnc.subgranule_offset): New members. Take care of the
case when the first audio buffer we get has a nonzero timestamp.
(GstVorbisEnc.next_ts): Renamed from prev_ts, because now we
properly timestamp vorbis buffers with the time of the first
sample, not the last.
* ext/vorbis/vorbisenc.c (granulepos_to_clocktime): Renamed from
vorbis_granule_time_copy -- now it takes the granule/subgranule
offset into account.
* tests/check/pipelines/vorbisenc.c: New test for correctness of
timestamps, durations, and granulepos on buffers produced by
vorbisenc.
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
Patch from Eric Jonas to support conversions to/from UYVY
(Fixes: #324626)
Original commit message from CVS:
* gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
* gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render)
use of gst_guint64_to_gdouble to be compliant with vs6
* gst/playback/gstdecodebin.c: (try_to_link_1)
* gst/videorate/videorate.c: (gst_video_rate_blank_data)
use of G_GINT64_CONSTANT for int64 constants
* win32/common/libgstinterfaces.def:
export some symbols (gst_mixer_get_type,gst_mixer_track_get_type)
* win32/vs6:
update and add new project files
Original commit message from CVS:
* Makefile.am:
* win32/MANIFEST:
* win32/common/interfaces-enumtypes.c:
(gst_color_balance_type_get_type), (gst_mixer_type_get_type),
(gst_mixer_track_flags_get_type),
(gst_tuner_channel_flags_get_type):
* win32/common/interfaces-enumtypes.h:
* win32/common/multichannel-enumtypes.c:
(gst_audio_channel_position_get_type):
* win32/common/multichannel-enumtypes.h:
add a win32-update rule like in core, and copy over enumtypes files
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_init), (set_hwparams),
(set_swparams), (gst_alsasink_prepare), (gst_alsasink_unprepare),
(gst_alsasink_close), (gst_alsasink_write), (gst_alsasink_reset):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (set_hwparams),
(set_swparams), (gst_alsasrc_open), (gst_alsasrc_prepare),
(gst_alsasrc_unprepare), (gst_alsasrc_read):
Update all error messages. All of them should either use
the default translated message, or actually provide a
translatable string.
Make the string for channel count problems meaningful.
Original commit message from CVS:
* gst/videoscale/vs_scanline.c: Oops, *that's* why I never
checked in this change -- it requires liboil features not
in 0.3.6. Revert parts.
Original commit message from CVS:
* ext/libvisual/visual.c: (get_buffer):
When pad_alloc returns a GstFlowReturn other
than GST_FLOW_OK, make sure it is passed upstream.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
(gst_alsasink_class_init):
Free the device name string.
* ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init),
(gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad),
(gst_ogg_mux_handle_src_event), (gst_ogg_mux_clear_collectpads):
Don't remove a pad from the collectpads structure until it
is released - it's a request pad, and may receive data again
if the element gets moved back to PLAYING state.
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
Ensure we turn on double buffering on the Xv port, and
set the colour key to something dark and mysterious that
isn't black.
Original commit message from CVS:
* ext/alsa/gstalsaplugin.c: (plugin_init):
* ext/cdparanoia/gstcdparanoiasrc.c:
(gst_cd_paranoia_src_base_init), (plugin_init):
* ext/gnomevfs/gstgnomevfs.c: (plugin_init):
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
- a library should not call setlocale. see Libraries node in
gettext manual
- make sure all plugins that use translation do bindtextdomain
to point to the localedir
* gst/playback/gstplaybin.c: (gen_vis_element), (add_sink),
(setup_sinks), (plugin_init):
all this, and check for NULL when creating sinks
Original commit message from CVS:
2006-01-27 Julien MOUTTE <julien@moutte.net>
* gst/subparse/gstsubparse.c: (gst_subparse_type_find),
(plugin_init): Make typefinding of subtitles work again.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (aac_type_find),
(mp3_type_frame_length_from_header), (mp3_type_find),
(wavpack_type_find), (m4a_type_find), (ircam_type_find),
(plugin_init):
Backport a bunch of typefinding fixes from the 0.8 branch.
Also, improve wavpack typefinding: if we can't peek the
entire wavpack block, try to parse the bits we can get and
see if we find what we're looking for in those.
Original commit message from CVS:
2006-01-26 Julien MOUTTE <julien@moutte.net>
* sys/ximage/ximagesink.c:
(gst_ximagesink_calculate_pixel_aspect_ratio):
* sys/xvimage/xvimagesink.c:
(gst_xvimagesink_calculate_pixel_aspect_ratio): Handle some
more cases of pixel aspect ratio.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (pad_probe):
Also consider the flush-start and tag events as unblockers
for the pad probes.
Original commit message from CVS:
2006-01-26 Julien MOUTTE <julien@moutte.net>
* gst/playback/gstplaybin.c: (gst_play_bin_init),
(gst_play_bin_dispose), (gst_play_bin_vis_unblocked),
(gst_play_bin_vis_blocked), (gst_play_bin_set_property):
On the fly visualisation switch, works disabling, enabling as
well but it won't be able to enable vis in a playbin that was
created with no visualisation.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Undo previous commit, it breaks resume after pause.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_setcaps), (gst_base_audio_sink_event),
(gst_base_audio_sink_preroll), (gst_base_audio_sink_render):
Improve debugging.
Post error when caps cannot be parsed.
Resync on discontinuity in the stream.
Clip samples to segment boundaries.
return WRONG_STATE sooner when we are flushing.
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
(gst_base_audio_src_get_time), (gst_base_audio_src_create):
Make audiosrc operate in TIME.
Set TIMESTAMP and DURATION on buffers.
