Commit graph

302 commits

Author SHA1 Message Date
Wim Taymans
a0354a5b96 gst-libs/gst/audio/gstbaseaudiosrc.c: Don't try to post an error message when setting the clock fails as this can hap...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_set_clock),
(gst_base_audio_src_check_get_range), (gst_base_audio_src_create):
Don't try to post an error message when setting the clock fails
as this can happen when adding an element to a bin which will then
deadlock. Fixes #347296.
2006-07-12 13:24:19 +00:00
Wim Taymans
ccee48bb85 Revert last two changes that broke the freeze.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
(gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
Revert last two changes that broke the freeze.
2006-07-12 11:28:37 +00:00
Wim Taymans
46d86d8005 gst-libs/gst/audio/gstringbuffer.c: Calculate correct silence samples so we don't fill our ringbuffer with noise.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
(gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
Calculate correct silence samples so we don't fill our ringbuffer
with noise.
2006-07-12 10:58:42 +00:00
Wim Taymans
fa5dacc998 gst-libs/gst/audio/gstbaseaudiosink.c: Use gobject_class instead of G_OBJECT_CLASS (klass)
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init),
(gst_base_audio_sink_provide_clock):
Use gobject_class instead of G_OBJECT_CLASS (klass)
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init), (gst_base_audio_src_init),
(gst_base_audio_src_set_clock), (gst_base_audio_src_provide_clock),
(gst_base_audio_src_get_time),
(gst_base_audio_src_check_get_range), (gst_base_audio_src_create),
(gst_base_audio_src_create_ringbuffer):
Fix latency and buffer-time constants and properties ala basesink.
Implement pull based scheduling. Fixes #346527.
Set default blocksize in GstBaseSrc to 0, we default to pushing out
one segment.
Refuse slaving to another clock instead of silently not working.
Only provide a clock when we are actually able to do so.
Various small cleanups and compiler hints.
2006-07-06 15:54:50 +00:00
Stefan Kost
cade791150 docs/libs/: add remaining symbols into correct setions
Original commit message from CVS:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
add remaining symbols into correct setions
* gst-libs/gst/audio/gstringbuffer.c:
fix incomplete docs
* gst-libs/gst/audio/gstringbuffer.h:
comment out not yet implemented function
* gst-libs/gst/floatcast/floatcast.h:
* gst-libs/gst/netbuffer/gstnetbuffer.c:
add short descriptions
* gst-libs/gst/interfaces/propertyprobe.c:
fix return value docs
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
simplify debug logging
* gst-libs/gst/riff/riff-read.h:
sync function prototype and docs
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
remove left over symbol
2006-06-16 10:02:25 +00:00
Thomas Vander Stichele
51ca8fe3e1 move last template doc snippets to source code and delete them
Original commit message from CVS:
* docs/libs/tmpl/gstaudio.sgml:
* docs/libs/tmpl/gstcolorbalance.sgml:
* docs/libs/tmpl/gstmixer.sgml:
* docs/libs/tmpl/gstringbuffer.sgml:
* docs/libs/tmpl/gsttuner.sgml:
* docs/libs/tmpl/gstxoverlay.sgml:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/gstringbuffer.c:
* gst-libs/gst/interfaces/colorbalance.c:
* gst-libs/gst/interfaces/mixer.c:
* gst-libs/gst/interfaces/tuner.c:
* gst-libs/gst/interfaces/xoverlay.c:
move last template doc snippets to source code and delete them
2006-06-07 11:03:03 +00:00
Jan Schmidt
45e06fe704 gst-libs/gst/audio/: Document better the fact that latency_time and buffer_time are values stored in microseconds, an...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_setcaps):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps),
(gst_ring_buffer_samples_done):
* gst-libs/gst/audio/gstringbuffer.h:
Document better the fact that latency_time and buffer_time are values
stored in microseconds, and not the usual GStreamer nanoseconds.
Change the variables (compatibly) that store them from GstClockTime
to guint64 to make it more clear that they're not storing clock times.
Also, remove the bogus property description that says the user can
specify -1 to get the default value, since that's never been the case.
When computing the default segment size for the ring buffer, make it
an integer number of samples.
When the sub-class indicates a delay greater than the number of
samples we've written return 0 from the audio sink get_time method.
2006-06-03 21:06:49 +00:00
Stefan Kost
131fb86b4b Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
Original commit message from CVS:
* ext/alsa/gstalsamixerelement.h:
* ext/alsa/gstalsamixeroptions.h:
* ext/alsa/gstalsamixertrack.h:
* ext/gnomevfs/gstgnomevfssink.h:
* ext/gnomevfs/gstgnomevfssrc.h:
* ext/theora/gsttheoradec.h:
* ext/theora/gsttheoraenc.h:
* ext/theora/gsttheoraparse.h:
* ext/vorbis/vorbisparse.h:
* gst-libs/gst/audio/gstaudioclock.h:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
* gst/audioconvert/gstaudioconvert.h:
* gst/audioresample/gstaudioresample.h:
* gst/audiotestsrc/gstaudiotestsrc.h:
* gst/ffmpegcolorspace/gstffmpegcolorspace.h:
* gst/playback/gststreamselector.h:
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcpclientsink.h:
* gst/tcp/gsttcpclientsrc.h:
* gst/tcp/gsttcpserversink.h:
* gst/tcp/gsttcpserversrc.h:
* gst/videorate/gstvideorate.h:
* gst/videoscale/gstvideoscale.h:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/volume/gstvolume.h:
* sys/v4l/gstv4ljpegsrc.h:
* sys/v4l/gstv4lmjpegsink.h:
* sys/v4l/gstv4lmjpegsrc.h:
* sys/v4l/gstv4lsrc.h:
* sys/ximage/ximagesink.h:
* sys/xvimage/xvimagesink.h:
* tests/old/testsuite/alsa/sinesrc.h:
Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
2006-06-01 19:19:51 +00:00
Tim-Philipp Müller
10d35563dd gst-libs/gst/audio/multichannel.c: It's okay to have caps with channels=1 and a channel position different from GST_A...
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions):
It's okay to have caps with channels=1 and a channel position
different from GST_AUDIO_CHANNEL_POSITION_FRONT_MONO
(deinterleavers might want to keep the position in the caps,
so that they can be re-interleaved again properly later).
Leave check for unexpected 2-channel layouts intact for now.
2006-05-16 17:34:14 +00:00
Stefan Kost
e972defd3e make GstElementDetails const
Original commit message from CVS:
* ext/alsa/gstalsamixerelement.c:
* ext/alsa/gstalsasrc.c:
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/gnomevfs/gstgnomevfssink.c:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c:
* ext/ogg/gstogmparse.c:
* ext/pango/gstclockoverlay.c:
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
* ext/pango/gsttimeoverlay.c:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* gst-libs/gst/audio/gstaudiofilter.c:
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
* gst/audioconvert/gstaudioconvert.c:
* gst/audiorate/gstaudiorate.c:
* gst/audioresample/gstaudioresample.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/playback/gstdecodebin.c:
* gst/playback/gstplaybin.c:
* gst/playback/gststreamselector.c:
* gst/subparse/gstsubparse.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gsttcpclientsink.c:
* gst/tcp/gsttcpclientsrc.c:
* gst/tcp/gsttcpserversink.c:
* gst/tcp/gsttcpserversrc.c:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
* gst/videorate/gstvideorate.c:
* gst/videoscale/gstvideoscale.c:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/volume/gstvolume.c:
* sys/v4l/gstv4ljpegsrc.c:
* sys/v4l/gstv4lmjpegsink.c:
* sys/v4l/gstv4lmjpegsrc.c:
* sys/v4l/gstv4lsrc.c:
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
* tests/check/libs/cddabasesrc.c:
make GstElementDetails const
2006-04-28 19:46:37 +00:00
Wim Taymans
102b79e46e gst-libs/gst/audio/gstbaseaudiosink.c: patch to make timestamp checking more tollerant to rounding errors given that ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_event),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
patch to make timestamp checking more tollerant to rounding
errors given that real discontinuities are to be marked on
buffers. Fixes some asf files and #338778.
Also avoid some crashers when we receive an event in the
NULL state.
2006-04-28 15:08:09 +00:00
Wim Taymans
04754176a6 gst-libs/gst/audio/gstringbuffer.c: Check arguments passed to public functions instead of crashing.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_set_callback), (gst_ring_buffer_acquire),
(gst_ring_buffer_release), (gst_ring_buffer_is_acquired),
(gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
(gst_ring_buffer_pause), (gst_ring_buffer_stop),
(gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
(gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
(gst_ring_buffer_commit), (gst_ring_buffer_read),
(gst_ring_buffer_prepare_read), (gst_ring_buffer_advance),
(gst_ring_buffer_clear), (gst_ring_buffer_may_start):
Check arguments passed to public functions instead of
crashing.
2006-04-28 14:48:11 +00:00
Wim Taymans
c068425b38 gst-libs/gst/audio/gstbaseaudiosrc.c: GstBaseAudioSrc must be live or it does not work.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
(gst_base_audio_src_get_time), (gst_base_audio_src_create):
GstBaseAudioSrc must be live or it does not work.
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audio_src_init):
Don't set live to TRUE as this is the default in the parentclass.
2006-04-28 14:37:46 +00:00
Wim Taymans
35058f78c1 gst-libs/gst/audio/gstbaseaudiosink.c: Starting the ringbuffer when we did not acquire it can cause a deadlock, is po...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event):
Starting the ringbuffer when we did not acquire it can cause
a deadlock, is pointless and causes nasty things for
subclasses.
Fixes gst-launch audiotestsrc num-buffers=0 ! alsasink.
2006-04-10 17:05:46 +00:00
Stefan Kost
0afac375b4 Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
Original commit message from CVS:
* ext/alsa/gstalsamixeroptions.c:
(gst_alsa_mixer_options_class_init):
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_class_init):
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
* ext/ogg/gstoggparse.c: (gst_ogg_parse_class_init):
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_class_init):
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_class_init):
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_class_init):
* gst-libs/gst/audio/gstaudiosrc.c:
(gst_audioringbuffer_class_init):
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_class_init):
* gst-libs/gst/interfaces/colorbalancechannel.c:
(gst_color_balance_channel_class_init):
* gst-libs/gst/interfaces/mixeroptions.c:
(gst_mixer_options_class_init):
* gst-libs/gst/interfaces/mixertrack.c:
(gst_mixer_track_class_init):
* gst-libs/gst/interfaces/tunerchannel.c:
(gst_tuner_channel_class_init):
* gst-libs/gst/interfaces/tunernorm.c: (gst_tuner_norm_class_init):
* gst-libs/gst/netbuffer/gstnetbuffer.c:
(gst_netbuffer_class_init):
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init):
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init):
* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
* gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
* gst/playback/gststreamselector.c:
(gst_stream_selector_class_init):
* gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
* sys/v4l/gstv4lcolorbalance.c:
(gst_v4l_color_balance_channel_class_init):
* sys/v4l/gstv4ljpegsrc.c: (gst_v4ljpegsrc_class_init):
* sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
* sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
* sys/v4l/gstv4ltuner.c: (gst_v4l_tuner_channel_class_init),
(gst_v4l_tuner_norm_class_init):
* sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
* tests/old/testsuite/alsa/sinesrc.c: (sinesrc_class_init):
Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
2006-04-08 21:02:53 +00:00
Stefan Kost
1a2642a1d2 Fix broken GObject macros
Original commit message from CVS:
* ext/pango/gsttextrender.h:
* gst-libs/gst/audio/gstaudiosink.h:
* gst-libs/gst/audio/gstaudiosrc.h:
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstringbuffer.h:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
* gst-libs/gst/rtp/gstbasertppayload.h:
* gst-libs/gst/video/gstvideofilter.h:
* gst-libs/gst/video/gstvideosink.h:
* gst/playback/gstplaybasebin.h:
* gst/tcp/gstmultifdsink.h:
* sys/v4l/gstv4lelement.h:
Fix broken GObject macros
2006-04-08 18:09:17 +00:00
Stefan Kost
2d826700fa Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init),
(gst_gnome_vfs_sink_class_init):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init):
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init):
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init):
* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
(gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init):
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_base_init):
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_base_init):
* gst/adder/gstadder.c: (gst_adder_get_type):
* gst/adder/gstadder.h:
* gst/audioconvert/gstaudioconvert.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
(gst_audio_test_src_create):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/playback/gstdecodebin.c:
* gst/playback/gstplaybin.c:
* gst/playback/gststreamselector.c:
(gst_stream_selector_base_init):
* gst/subparse/gstsubparse.c: (gst_sub_parse_base_init):
* gst/volume/gstvolume.c:
* sys/v4l/gstv4lmjpegsink.c:
* sys/v4l/gstv4lmjpegsrc.c:
* tests/check/libs/cddabasesrc.c:
* tests/old/examples/gob/gst-identity2.gob:
Add docs for adder, use GST_ELEMENT_DETAILS macro,
define GstElementDetails at the top
2006-03-24 10:42:11 +00:00
Wim Taymans
4df07064b8 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix audio sources, forgot to make the ringbuffer startable...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_get_time), (gst_base_audio_src_create),
(gst_base_audio_src_change_state):
Fix audio sources, forgot to make the ringbuffer
startable...
2006-03-23 16:58:03 +00:00
Wim Taymans
2df1088b3f gst-libs/gst/audio/gstbaseaudiosrc.c: unparent instead of unref the ringbuffer.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_get_time), (gst_base_audio_src_create),
(gst_base_audio_src_change_state):
unparent instead of unref the ringbuffer.
2006-03-23 16:29:58 +00:00
Wim Taymans
227474e464 gst-libs/gst/audio/gstbaseaudiosink.c: Implement new async_play vmethod to start slaving and allow playback start in ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_async_play),
(gst_base_audio_sink_do_play), (gst_base_audio_sink_change_state):
Implement new async_play vmethod to start slaving and allow
playback start in case of async PLAY state changes.
* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
Enable QoS with new method in base class.
2006-03-23 16:24:23 +00:00
Wim Taymans
747d560fb5 gst-libs/gst/audio/gstbaseaudiosink.c: Since we _parent the ringbuffer, we also need to _unparent instead of a plain ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_dispose):
Since we _parent the ringbuffer, we also need to
_unparent instead of a plain _unref.
2006-03-22 12:33:09 +00:00
Wim Taymans
82fd38fbcf gst-libs/gst/audio/: Only start playback if we are playing. should fix #330748.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstringbuffer.c: (wait_segment),
(gst_ring_buffer_may_start):
* gst-libs/gst/audio/gstringbuffer.h:
Only start playback if we are playing.
should fix #330748.
2006-03-17 17:48:33 +00:00
Tim-Philipp Müller
ab6f99ab60 gst-libs/gst/audio/gstaudiofilter.c: Don't ignore flow return from gst_pad_push().
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_chain):
Don't ignore flow return from gst_pad_push().
2006-03-07 13:01:21 +00:00
Christophe Fergeau
8e6d3a5c03 Don't leak references returned by gst_pad_get_parent()
Original commit message from CVS:
* ext/libvisual/visual.c: (gst_visual_getcaps),
(gst_visual_src_setcaps), (gst_visual_sink_setcaps):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_sinkconnect):
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
(gst_vorbisenc_convert_sink):
* gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
(gst_audio_duration_from_pad_buffer):
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_link),
(gst_audio_filter_chain):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps):
* gst-libs/gst/video/video.c: (gst_video_frame_rate),
(gst_video_get_size):
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
Don't leak references returned by gst_pad_get_parent()
(#333663, based on patch by: Christophe Fergeau).
2006-03-07 12:49:03 +00:00
Wim Taymans
1e9f5c43ad docs/: Added some more docs to libs and plugins.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
Added some more docs to libs and plugins.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_prepare_read), (gst_ring_buffer_clear):
* gst-libs/gst/audio/gstringbuffer.h:
Document ringbuffer some more.
* gst/videorate/gstvideorate.c: (gst_video_rate_class_init),
(gst_video_rate_setcaps), (gst_video_rate_reset),
(gst_video_rate_init), (gst_video_rate_flush_prev),
(gst_video_rate_swap_prev), (gst_video_rate_event),
(gst_video_rate_chain), (gst_video_rate_change_state):
* gst/videorate/gstvideorate.h:
Fix videorate to use segments.
Make it work with 0/1 framerates (closes #331903)
Handle EOS correctly.
Added docs.
2006-03-02 16:47:34 +00:00
Wim Taymans
77ff8c9fdb gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock in the NULL state.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_provide_clock):
Don't try to provide a clock in the NULL state.
2006-02-28 11:06:24 +00:00
Tim-Philipp Müller
043c6d91df gst-libs/gst/audio/mixerutils.c: Make order in which elements are tried more determinable.
Original commit message from CVS:
* gst-libs/gst/audio/mixerutils.c:
(element_factory_rank_compare_func):
Make order in which elements are tried more determinable.
2006-02-20 16:21:14 +00:00
Wim Taymans
3451a81879 gst-libs/gst/audio/gstringbuffer.c: Small cleanups.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_get_type),
(gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
(gst_ring_buffer_device_is_open), (gst_ring_buffer_acquire),
(gst_ring_buffer_release), (gst_ring_buffer_set_flushing),
(gst_ring_buffer_start), (gst_ring_buffer_pause_unlocked),
(gst_ring_buffer_pause), (gst_ring_buffer_stop),
(gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
(gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
(gst_ring_buffer_commit), (gst_ring_buffer_prepare_read),
(gst_ring_buffer_clear):
Small cleanups.
Added some G_LIKELY.
2006-02-17 14:07:01 +00:00
Wim Taymans
454618e9b9 gst-libs/gst/audio/TODO: Update TODO
Original commit message from CVS:
* gst-libs/gst/audio/TODO:
Update TODO

* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_offset):
When trying to play samples ASAP and we don't have a
previous sample, try to play at position 0 instead of
an invalid position.
2006-02-17 10:15:52 +00:00
Tim-Philipp Müller
9490d413c0 gst-libs/gst/audio/multichannel.c: Minor docs fix.
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
Minor docs fix.
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_wavext_add_channel_layout), (gst_riff_create_audio_caps):
Add support for WAVEFORMATEX, eg. PCM audio with more than two
channels and a channel layout map.
2006-02-16 19:18:46 +00:00
Tim-Philipp Müller
5b788a8a66 gst-libs/gst/audio/multichannel.c: When we have more than 2 channels, but no channel layout is specified in the caps,...
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_get_channel_positions):
When we have more than 2 channels, but no channel layout is
specified in the caps, return some default channel layout
to the caller and warn about about a possibly buggy element
(could be buggy filtercaps as well of course) (#317038).
2006-02-16 11:44:43 +00:00
Wim Taymans
3b45740289 gst-libs/gst/audio/gstringbuffer.c: Add some compiler G_(UN_)LIKELY help.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_pause_unlocked), (gst_ring_buffer_delay),
(gst_ring_buffer_samples_done), (wait_segment),
(gst_ring_buffer_commit), (gst_ring_buffer_clear):
Add some compiler G_(UN_)LIKELY help.
SIGNAL the ringbuffer waiters when going to PAUSED as well to
make sure they can exit their functions. Should fix #330748
2006-02-14 13:45:35 +00:00
Wim Taymans
16dbdc5c21 gst-libs/gst/audio/gstbaseaudiosink.c: Always sync on first sample we receive when starting.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_time), (gst_base_audio_sink_preroll),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Always sync on first sample we receive when starting.
2006-02-13 18:49:02 +00:00
Wim Taymans
0be7d56eb9 gst-libs/gst/audio/gstbaseaudiosink.c: Use scale functions when possible.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_time), (gst_base_audio_sink_setcaps),
(gst_base_audio_sink_drain), (gst_base_audio_sink_preroll),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Use scale functions when possible.
Fix error messages.
Free clockid when after waiting for EOS.
Use G_(UN_)LIKLY when it makes sense.
Fix sample clipping bug found by Arwed v. Merkatz fixes #330789.
2006-02-12 14:54:55 +00:00
Andy Wingo
4e0c846fa4 kapowpowpow
Original commit message from CVS:
kapowpowpow
2006-02-09 11:46:03 +00:00
Andy Wingo
4ae63e7361 gst-libs/gst/audio/gstringbuffer.c
Original commit message from CVS:
2006-02-09  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstringbuffer.c
(gst_ring_buffer_samples_done): Cast to guint64, fixes an integer
overflow after 13.5 hours of recording. Kapow!

* ext/alsa/gstalsasrc.c (gst_alsasrc_delay): Clamp the delay to
the buffer size -- we don't care about underrun/overrun reporting
right now, just need to return a useful value.
2006-02-09 11:36:18 +00:00
Wim Taymans
260b5295c9 gst-libs/gst/audio/gstbaseaudiosink.c: Ugh.. getting late I guess...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_provide_clock):
Ugh.. getting late I guess...
2006-02-02 18:18:31 +00:00
Wim Taymans
c78a5d7e1e gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock when we are not negotiated since we might not be ...
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.
2006-02-02 18:13:26 +00:00
Wim Taymans
416c011f11 gst-libs/gst/audio/TODO: Updated.
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.
2006-02-02 12:14:35 +00:00
Wim Taymans
a169abc679 gst-libs/gst/audio/gstaudiosink.c: Implement pause that does not wait for completion.
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.
2006-01-30 16:19:33 +00:00
Sébastien Moutte
dc46970cdf gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
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
2006-01-29 19:13:39 +00:00
Tim-Philipp Müller
27ed152e10 gst-libs/gst/audio/gstringbuffer.c: Make gcc-4.1 happy (part of #327357).
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format):
Make gcc-4.1 happy (part of #327357).
2006-01-28 18:19:18 +00:00
Wim Taymans
ccd05fa086 gst-libs/gst/audio/gstbaseaudiosink.c: Undo previous commit, it breaks resume after pause.
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.
2006-01-25 10:50:32 +00:00
Wim Taymans
2bc5ca1786 gst-libs/gst/audio/gstbaseaudiosink.c: Improve debugging.
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.
2006-01-25 09:27:01 +00:00
Jan Schmidt
04333a568c gst-libs/gst/audio/gstbaseaudiosink.c: Fix playback of non-synchronised streams by assuming a rate of 1.0 instead of ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Fix playback of non-synchronised streams by assuming a rate
of 1.0 instead of a random one.

Makes this work again:

gst-launch filesrc location=raw_audio.file ! 'audio/x-raw-int,
endianness=(int)4321, signed=(boolean)true, width=(int)16,
depth=(int)16, rate=(int)44100, channels=(int)2' ! audioconvert !
audioresample ! alsasink
2006-01-17 11:43:49 +00:00
Tim-Philipp Müller
f220f8295b Add docs for mixerutils stuff.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/mixerutils.c:
* gst-libs/gst/audio/mixerutils.h:
Add docs for mixerutils stuff.
2006-01-14 12:52:22 +00:00
Thomas Vander Stichele
5fd8ee2ea4 gst-libs/gst/audio/mixerutils.c: actually save the element we create
Original commit message from CVS:

* gst-libs/gst/audio/mixerutils.c:
(gst_audio_mixer_filter_do_filter):
actually save the element we create
2006-01-13 16:45:50 +00:00
Tim-Philipp Müller
b867510721 gst-libs/gst/audio/gstringbuffer.c: Set depth and width for alaw/mulaw (fixes #326601).
Original commit message from CVS:
Reviewed by: Tim-Philipp Müller  <tim at centricular dot net>
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
Set depth and width for alaw/mulaw (fixes #326601).
2006-01-11 15:11:20 +00:00
Michael Smith
b0c21cab17 gst-libs/gst/audio/gstaudiosrc.c: Don't leak GCond in audio sources.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_dispose):
Don't leak GCond in audio sources.
2006-01-10 12:25:59 +00:00
Tim-Philipp Müller
8ec22e812b gst-libs/gst/audio/gstringbuffer.*: Name (private) union, makes Forte compiler happy (this time for real) (#324900).
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
(gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
(gst_ring_buffer_pause), (wait_segment):
* gst-libs/gst/audio/gstringbuffer.h:
Name (private) union, makes Forte compiler happy (this time
for real) (#324900).
2006-01-10 09:38:44 +00:00
Tim-Philipp Müller
3b96467f63 gst-libs/gst/audio/Makefile.am: Link against libgstinterfaces, needed for mixer and property probe stuff.
Original commit message from CVS:
* gst-libs/gst/audio/Makefile.am:
Link against libgstinterfaces, needed for mixer
and property probe stuff.
2006-01-09 10:52:33 +00:00
Tim-Philipp Müller
e737f441d3 gst-libs/gst/audio/: Add gst_audio_default_registry_mixer_filter() utility function.
Original commit message from CVS:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/mixerutils.c:
(gst_audio_mixer_filter_do_filter),
(gst_audio_mixer_filter_check_element),
(gst_audio_mixer_filter_probe_feature),
(element_factory_rank_compare_func),
(gst_audio_default_registry_mixer_filter):
* gst-libs/gst/audio/mixerutils.h:
Add gst_audio_default_registry_mixer_filter() utility
function.
2006-01-09 09:38:34 +00:00
Tim-Philipp Müller
be8f055317 gst-libs/gst/audio/gstringbuffer.*: Sun's Forte compiler doesn't seem to like anonymous structs, so use same setup as...
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
(gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
(gst_ring_buffer_pause), (wait_segment):
* gst-libs/gst/audio/gstringbuffer.h:
Sun's Forte compiler doesn't seem to like anonymous structs,
so use same setup as in GstBaseSrc (fixes #324900).
2006-01-02 23:37:38 +00:00
Thomas Vander Stichele
01bc68f918 stop making fun of older compilers
Original commit message from CVS:
stop making fun of older compilers
2005-12-20 12:24:29 +00:00
Thomas Vander Stichele
b4b2b62a74 gst-libs/gst/audio/: update strings, values are in microseconds change the default sink buffer time to something that...
Original commit message from CVS:

* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
update strings, values are in microseconds
change the default sink buffer time to something that is smaller
(to help software volume mixing have a slightly lower delay) but
still be acceptable on Wim's laptop
2005-12-20 12:00:26 +00:00
Thomas Vander Stichele
5f83aa7dfa expand tabs
Original commit message from CVS:
expand tabs
2005-12-06 19:42:02 +00:00
Thomas Vander Stichele
9db2e7681a borgify
Original commit message from CVS:
borgify
2005-12-01 14:29:09 +00:00
Thomas Vander Stichele
8823933bcd folded audiofilter into the audio library
Original commit message from CVS:
folded audiofilter into the audio library
2005-11-29 01:25:31 +00:00
Wim Taymans
3f05db1828 gst-libs/gst/audio/TODO: Updated TODO
Original commit message from CVS:
* gst-libs/gst/audio/TODO:
Updated TODO

* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_open_device),
(gst_audioringbuffer_close_device), (gst_audioringbuffer_acquire),
(gst_audioringbuffer_release):
Small cleanups.

* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
Slave to the master clock when going to PLAYING and unslave when
going to PAUSED.

* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
(gst_ring_buffer_acquire), (gst_ring_buffer_release),
(gst_ring_buffer_samples_done), (gst_ring_buffer_set_sample),
(gst_ring_buffer_clear_all), (wait_segment),
(gst_ring_buffer_commit), (gst_ring_buffer_read),
(gst_ring_buffer_advance):
* gst-libs/gst/audio/gstringbuffer.h:
Add some docs and cleanups.
2005-11-28 15:53:55 +00:00
Thomas Vander Stichele
efb938bd9a configure.ac: added GST_LIB_LDFLAGS and GST_ALL_LDFLAGS
Original commit message from CVS:

* configure.ac:
added GST_LIB_LDFLAGS and GST_ALL_LDFLAGS
* gst-libs/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/net/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
and use them
2005-11-27 16:18:50 +00:00
Wim Taymans
c7dc33e495 gst-libs/gst/audio/gstringbuffer.c: If we are reading too slowly, jump forward in the ringbuffer instead of blocking.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_read):
If we are reading too slowly, jump forward in the ringbuffer
instead of blocking.
2005-11-23 13:29:50 +00:00
Wim Taymans
67b21a9033 gst-libs/gst/audio/gstbaseaudiosink.c: Fix for calibration API change.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_set_clock),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Fix for calibration API change.
2005-11-23 13:08:54 +00:00
Michael Smith
71f3969208 gst-libs/gst/audio/multichannel.c: Use gst_value_array_*() functions on value arrays, not gst_value_list_*().
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_get_channel_positions),
(gst_audio_set_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(gst_audio_fixate_channel_positions):
Use gst_value_array_*() functions on value arrays, not
gst_value_list_*().
2005-11-23 12:40:04 +00:00
Wim Taymans
af2acb6eea gst-libs/gst/audio/gstbaseaudiosink.c: And we provide a clock by default, of course...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_set_clock),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
And we provide a clock by default, of course...
2005-11-22 18:54:56 +00:00
Wim Taymans
a3cb4d4937 gst-libs/gst/audio/gstaudioclock.c: This clock can be slaved to a master clock now.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init):
This clock can be slaved to a master clock now.

* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
(gst_base_audio_sink_dispose), (gst_base_audio_sink_provide_clock),
(gst_base_audio_sink_set_clock),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_preroll),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Handle slaving the internal clock to the clock selected in the
pipeline.
Add property to make the basesink not provide a clock.

* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
(gst_base_rtp_depayload_wait):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
We can use the clock in GstElement, no need to store it ourselves.
2005-11-22 18:32:09 +00:00
Thomas Vander Stichele
08cd3b973f remove some deprecated functions
Original commit message from CVS:
remove some deprecated functions
2005-11-22 13:14:07 +00:00
Thomas Vander Stichele
1c3b6d42a9 gst-libs/gst/audio/audio.*: fix prototype - wondering why the test worked regardless
Original commit message from CVS:

* gst-libs/gst/audio/audio.c: (gst_audio_duration_from_pad_buffer):
* gst-libs/gst/audio/audio.h:
fix prototype - wondering why the test worked regardless
2005-11-21 23:51:45 +00:00
Thomas Vander Stichele
be5a7cd625 add a method that returns a proper GstClockTime
Original commit message from CVS:
add a method that returns a proper GstClockTime
2005-11-21 22:56:33 +00:00
Wim Taymans
0f2336cff6 gst/: Segment update fix.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
(gst_base_audio_sink_provide_clock),
(gst_base_audio_sink_set_clock), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
* gst/audioresample/gstaudioresample.c:
Segment update fix.
2005-11-21 17:14:02 +00:00
Andy Wingo
f405e12b4a *.*: Ran scripts/update-macros. Oh yes.
Original commit message from CVS:
2005-11-21  Andy Wingo  <wingo@pobox.com>

* *.h:
* *.c: Ran scripts/update-macros. Oh yes.
2005-11-21 16:35:24 +00:00
Jan Schmidt
1cc82e9138 Rename gst_caps_structure_fixate_* to gst_structure_fixate_* (#322027)
Original commit message from CVS:
* ext/libvisual/visual.c: (get_buffer):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_fixate):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_caps):
* gst/audioscale/gstaudioscale.c: (gst_audioscale_fixate):
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiotestsrc_src_fixate):
* gst/sine/gstsinesrc.c: (gst_sinesrc_src_fixate):
* gst/videorate/gstvideorate.c: (gst_videorate_setcaps):
* gst/videoscale/gstvideoscale.c: (gst_videoscale_fixate_caps):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_videotestsrc_src_fixate):
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_fixate):
Rename gst_caps_structure_fixate_* to gst_structure_fixate_*
(#322027)
2005-11-21 14:29:53 +00:00
Wim Taymans
9edbf81fd2 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix the audiosrc base class again, we did not unflush.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_change_state):
Fix the audiosrc base class again, we did not unflush.
2005-11-17 14:40:12 +00:00
Wim Taymans
99fb91493e gst-libs/gst/audio/gstbaseaudiosink.c: Set ringbuffer to non-flushing when going to PAUSED, set to flushing again whe...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_provide_clock),
(gst_base_audio_sink_change_state):
Set ringbuffer to non-flushing when going to PAUSED, set to
flushing again when going to READY.

* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
(gst_ring_buffer_stop):
Start in flushing mode by default.
Don't set flushing in the _stop method, let the app call
this explicitly.
2005-11-16 16:48:35 +00:00
Wim Taymans
8360581332 gst-libs/gst/audio/gstringbuffer.c: Set ringbuffer to flushing when stopping so that we don't block on wait_segment a...
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_stop):
Set ringbuffer to flushing when stopping so that we don't
block on wait_segment anymore and livelock.
2005-11-16 12:17:06 +00:00
Wim Taymans
b886b99345 gst-libs/gst/audio/gstbaseaudiosink.c: No need to do a typecheck.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
No need to do a typecheck.
2005-11-08 11:41:52 +00:00
Wim Taymans
d23d907a86 gst-libs/gst/audio/gstringbuffer.h: Don't break ABI.
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.
2005-10-31 11:43:01 +00:00
Wim Taymans
09ca2ec93b gst-libs/gst/audio/: Add flushing mode to the ringbuffer so that it in all cases does not try to handle more audio. T...
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.
2005-10-31 10:30:41 +00:00
Wim Taymans
a878cbdfe1 gst-libs/gst/audio/gstbaseaudiosink.c: Remove g_print
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.
2005-10-24 14:59:55 +00:00
Wim Taymans
cfadd55297 gst-libs/gst/audio/gstbaseaudiosink.c: Buffers with no timestamps get aligned with previous buffers or on underrun, p...
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.
2005-10-24 14:52:22 +00:00
Wim Taymans
7879080357 ext/: Fix old naming.
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.
2005-10-21 15:14:36 +00:00
Wim Taymans
fc8ce00673 Bye bye buffer-frames.
Original commit message from CVS:
* check/elements/audioconvert.c:
* docs/libs/tmpl/gstaudio.sgml:
* docs/libs/tmpl/gstcolorbalance.sgml:
* ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
(vorbis_handle_identification_packet), (vorbis_handle_data_packet):
* ext/vorbis/vorbisenc.c: (raw_caps_factory):
* gst-libs/gst/audio/audio.c: (gst_audio_structure_set_int):
* gst-libs/gst/audio/audio.h:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps):
* gst/volume/gstvolume.c:
Bye bye buffer-frames.
2005-10-19 17:02:46 +00:00
Wim Taymans
efb6fcb802 ext/alsa/gstalsasink.c: Set handle to NULL.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_init),
(gst_alsasink_close):
Set handle to NULL.

* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
(gst_ring_buffer_acquire), (gst_ring_buffer_release),
(gst_ring_buffer_start), (gst_ring_buffer_pause),
(gst_ring_buffer_stop), (gst_ring_buffer_commit),
(gst_ring_buffer_read):
More debug info.
2005-10-18 11:07:26 +00:00
Thomas Vander Stichele
4f8f42b0b6 restructure configure.ac, use correct libtool LDFLAGS, fix up defines
Original commit message from CVS:
restructure configure.ac, use correct libtool LDFLAGS, fix up defines
2005-10-16 13:54:44 +00:00
Wim Taymans
1355459057 gst-libs/gst/audio/gstringbuffer.c: Don't assert on normal stuff.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
(gst_ring_buffer_read), (gst_ring_buffer_clear):
Don't assert on normal stuff.

* gst/playback/gstplaybin.c: (do_playbin_seek):
API fix.
2005-10-12 12:38:20 +00:00
Wim Taymans
5c17d94013 gst-libs/gst/audio/: Cleanups.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
(gst_ring_buffer_read):
Cleanups.
Commit and read from ringbuffer in samples rather than bytes.
2005-10-11 18:32:01 +00:00
Wim Taymans
0c71c6348f gst-libs/gst/audio/gstbaseaudiosink.c: Respect segment rate and accum when scheduling samples.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Respect segment rate and accum when scheduling samples.
2005-10-11 17:31:48 +00:00
Wim Taymans
81a09fc472 ext/alsa/gstalsasink.c: Also allow unsigned int.
Original commit message from CVS:
* ext/alsa/gstalsasink.c:
Also allow unsigned int.

* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Small cleanup
2005-10-10 17:04:24 +00:00
Wim Taymans
d920233a73 gst-libs/gst/audio/gstaudiosink.c: Only actually wait for the thread to be stopped if it's running.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_stop):
Only actually wait for the thread to be stopped if it's
running.
2005-10-08 12:02:08 +00:00
Wim Taymans
bd80afd2d1 gst-libs/gst/audio/gstbaseaudiosink.c: If we receive EOS we can start playback of what we had.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render):
If we receive EOS we can start playback of what we had.
2005-10-08 11:47:52 +00:00
Edgard Lima
e846919fe9 gst-libs/gst/audio/: Fixed gstbaseaudiosrc adding ring buffer sync to it.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_event),
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_read):
patch from Edgard Lima <edgard.lima@indt.org.br>
Fixed gstbaseaudiosrc adding ring buffer sync to it.
2005-10-06 15:15:04 +00:00
Wim Taymans
a872aac9f8 ext/ogg/gstoggdemux.c: Report the FLOW_RETURN as string in the error message.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_loop):
Report the FLOW_RETURN as string in the error message.

* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_clear_all):
Don't assert when clearing an unnegotiated buffer.
2005-10-06 13:11:55 +00:00
Andy Wingo
721c97d438 gst-libs/gst/audio/gstringbuffer.c (gst_ring_buffer_clear)
Original commit message from CVS:
2005-10-02  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstringbuffer.c (gst_ring_buffer_clear)
(gst_ring_buffer_prepare_read):
* gst-libs/gst/audio/gstaudiosink.c (audioringbuffer_thread_func):
Demote to LOG.
2005-10-02 15:58:57 +00:00
Wim Taymans
79be8760f0 gst-libs/gst/audio/: get_clock -> provide_clock
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init),
(gst_base_audio_sink_provide_clock), (gst_base_audio_sink_render):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init),
(gst_base_audio_src_provide_clock):
get_clock -> provide_clock
2005-09-28 13:41:29 +00:00
Wim Taymans
b17856db22 Fix sync again. Moved sample alignment to basesink.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_reset):
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_class_init), (audioringbuffer_thread_func),
(gst_audioringbuffer_stop):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_stop),
(gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
(gst_ring_buffer_commit), (gst_ring_buffer_read):
* gst-libs/gst/audio/gstringbuffer.h:
Fix sync again. Moved sample alignment to basesink.
2005-09-24 13:06:03 +00:00
Thomas Vander Stichele
272aad79bb add/fix docs
Original commit message from CVS:

* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* gst/volume/gstvolume.c:
add/fix docs
* gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size):
* gst-libs/gst/audio/audio.h:
add conversion macros for frames <-> clocktime
2005-09-23 18:14:54 +00:00
David Schleef
cb8927cb92 Fixes for changes in registry API.
Original commit message from CVS:
* check/generic/states.c:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
* gst/playback/gstdecodebin.c: (gst_decode_bin_init):
Fixes for changes in registry API.
* configure.ac: Only export gst_plugins_desc.  Add -no-undefined
to GST_PLUGIN_LDFLAGS.
* ext/libvisual/visual.c: Make the library shut up.
* gst-libs/gst/audio/audio.c: Don't define a plugin in a library.
* gst-libs/gst/audio/gstaudiofilter.c: same
2005-09-15 06:59:36 +00:00
Stefan Kost
1aa698fefa gst/: fixing lost sync, some more debugging
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit):
* gst/volume/gstvolume.c: (gst_volume_class_init),
(volume_transform):
fixing lost sync, some more debugging
2005-09-08 22:42:11 +00:00
Andy Wingo
6665c3084c All plugins updated for element state changes.
Original commit message from CVS:
2005-09-02  Andy Wingo  <wingo@pobox.com>

* All plugins updated for element state changes.
2005-09-02 15:43:18 +00:00
Wim Taymans
44cc3421a0 gst-libs/gst/audio/gstbaseaudiosink.c: Resync if the buffer timestamps drift more than a 10th of a second.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Resync if the buffer timestamps drift more than a 10th
of a second.
2005-08-31 10:57:35 +00:00
Andy Wingo
c32721723b Updates for two-arg init from GST_BOILERPLATE_FULL.
Original commit message from CVS:
2005-08-28  Andy Wingo  <wingo@pobox.com>

* Updates for two-arg init from GST_BOILERPLATE_FULL.
2005-08-28 17:52:45 +00:00
Wim Taymans
7824216cef ext/ogg/gstoggdemux.c: Parse seeking events better.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
(gst_ogg_pad_event), (gst_ogg_demux_factory_filter),
(gst_ogg_pad_submit_packet), (gst_ogg_chain_new),
(gst_ogg_demux_init), (gst_ogg_demux_perform_seek),
(gst_ogg_demux_collect_chain_info), (gst_ogg_demux_collect_info),
(gst_ogg_demux_chain), (gst_ogg_demux_loop), (gst_ogg_print):
Parse seeking events better.
Unref static caps.
Generate correct newsegment events, fixes seeking in live oggs.

* ext/theora/theoradec.c: (theora_dec_src_query),
(theora_dec_src_event), (theora_dec_src_getcaps),
(theora_dec_sink_event), (theora_dec_push), (theora_dec_chain):
Use newsegment values to report correct play time.

* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
(vorbis_dec_src_event), (vorbis_dec_sink_event):
* ext/vorbis/vorbisdec.h:
Parse and use newsegment values to report correct play time.

* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render):
Clear ringbuffer on flush.
Use newsegment values to calculate playback time.

* sys/ximage/ximagesink.c: (gst_ximagesink_get_times):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_times):
Basesink does newsegment calculations for us now.
2005-08-24 18:04:45 +00:00
Wim Taymans
5ac2327f05 gst-libs/gst/audio/gstringbuffer.*: Added function to clear the ringbuffer.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_set_sample),
(gst_ring_buffer_clear_all):
* gst-libs/gst/audio/gstringbuffer.h:
Added function to clear the ringbuffer.
2005-08-24 11:29:10 +00:00
Andy Wingo
7afb104567 gst-libs/gst/audio/gstbaseaudiosrc.c
Original commit message from CVS:
2005-08-23  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstbaseaudiosrc.c
(gst_base_audio_src_fixate): Only fixate endianness if it is
present in the caps.
2005-08-23 13:29:17 +00:00
Andy Wingo
13b122a106 gst-libs/gst/audio/gstaudiosrc.*: Implement open_device and close_device in the ring buffer, like gstaudiosink.
Original commit message from CVS:
2005-08-22  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstaudiosrc.h:
* gst-libs/gst/audio/gstaudiosrc.c: Implement open_device and
close_device in the ring buffer, like gstaudiosink.

* ext/alsa/gstalsamixer.h:
* ext/alsa/gstalsamixer.c: Not a GObject any more. Include a nifty
macro to implement the interface without much code. Cleanups.

* ext/alsa/gstalsasrc.h:
* ext/alsa/gstalsasrc.c: Be a mixer. Open device and mixer in
READY.

* ext/alsa/Makefile.am: Add new files.
* ext/alsa/gstalsamixerelement.c:
* ext/alsa/gstalsamixerelement.c: Split element code out from
mixer code so that alsasrc can be a mixer too.
2005-08-22 15:11:31 +00:00
Wim Taymans
4e3b19e5fb gst-libs/gst/audio/gstbaseaudiosrc.c: Open and close device in READY<->NULL state change.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_change_state):
Open and close device in READY<->NULL state change.
2005-08-16 15:53:59 +00:00
Tim-Philipp Müller
b9b56ce7d3 gst-libs/gst/: Add padding (you will need to rebuild gst-plugins-base, gst-plugins and all applications afterwards!)
Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.h:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/audio/gstaudiosink.h:
* gst-libs/gst/audio/gstaudiosrc.h:
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstringbuffer.h:
* gst-libs/gst/net/gstnetbuffer.h:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
* gst-libs/gst/rtp/gstrtpbuffer.h:
Add padding (you will need to rebuild gst-plugins-base,
gst-plugins and all applications afterwards!)
2005-08-09 17:29:40 +00:00
Andy Wingo
69d36f02ce gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
2005-08-08  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_change_state): Open the device in NULL->READY
like good elements should. Close on READY->NULL too.

* gst-libs/gst/audio/gstaudiosink.c
(gst_audioringbuffer_open_device,
(gst_audioringbuffer_close_device, gst_audioringbuffer_acquire)
(gst_audioringbuffer_release): Updates for new ring buffer API,
hook into the new audio sink api.

* gst-libs/gst/audio/gstaudiosink.h (GstAudioSinkClass.open)
(GstAudioSinkClass.close): Just open and close the device -- no
resource allocation or configuration.
(GstAudioSinkClass.prepare, GstAudioSinkClass.unprepare): New
vmethods, handle device setup and resource allocation.

* ext/alsa/gstalsasink.c (gst_alsasink_open, gst_alsasink_close)
(gst_alsasink_prepare, gst_alsasink_unprepare): Update for new
base class API.

* gst-libs/gst/audio/gstringbuffer.h
(GstRingBufferClass.open_device, GstRingBufferClass.close_device):
New vmethods.

* gst-libs/gst/audio/gstringbuffer.c (gst_ring_buffer_open_device)
(gst_ring_buffer_close_device, gst_ring_buffer_device_is_open):
New API functions. The device should be opened before acquiring
and closed after releasing.
2005-08-08 16:42:10 +00:00
Wim Taymans
78b9a84efa gst-libs/gst/audio/gstbaseaudiosrc.c: More compilation fixen.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_event):
More compilation fixen.
2005-07-27 19:13:27 +00:00
Wim Taymans
50b9b8acc4 gst-libs/gst/audio/gstbaseaudiosink.c: Fix compilation.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render),
(gst_base_audio_sink_create_ringbuffer),
(gst_base_audio_sink_change_state):
Fix compilation.
2005-07-27 19:10:20 +00:00
Wim Taymans
e2da9961d9 ext/ogg/gstoggdemux.c: Generate correct disconts for live chained oggs.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_event),
(gst_ogg_pad_internal_chain), (gst_ogg_pad_typefind),
(gst_ogg_demux_chain_elem_pad), (gst_ogg_demux_queue_data),
(gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
(gst_ogg_pad_submit_page), (gst_ogg_chain_new),
(gst_ogg_demux_init), (gst_ogg_demux_activate_chain),
(gst_ogg_demux_perform_seek), (gst_ogg_demux_collect_chain_info),
(gst_ogg_demux_collect_info), (gst_ogg_demux_chain),
(gst_ogg_demux_send_event), (gst_ogg_demux_loop):
Generate correct disconts for live chained oggs.

* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render),
(gst_base_audio_sink_create_ringbuffer),
(gst_base_audio_sink_change_state):
Handle discont math correctly.

* gst/playback/gstplaybin.c: (add_sink):
Some small debug cleanup.
2005-07-21 17:25:40 +00:00
Ronald S. Bultje
7795794bf0 Fixes for API changes in core.
Original commit message from CVS:
* ext/ogg/gstoggmux.c: (gst_ogg_mux_get_headers),
(gst_ogg_mux_set_header_on_caps):
* ext/theora/theoraenc.c: (theora_set_header_on_caps):
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_set_header_on_caps):
* ext/vorbis/vorbisparse.c: (vorbis_parse_set_header_on_caps):
* gst-libs/gst/audio/multichannel.c:
(gst_audio_set_channel_positions),
(gst_audio_set_structure_channel_positions_list):
* gst/playback/gstdecodebin.c: (dynamic_create):
* gst/playback/gstplaybasebin.c: (setup_source), (mute_group_type):
* gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
Fixes for API changes in core.
2005-07-20 17:16:54 +00:00
Wim Taymans
ee345636bc gst-libs/gst/audio/: Make sure the audio clock always returns an increasing value.
Original commit message from CVS:
* gst-libs/gst/audio/TODO:
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
(gst_audio_clock_get_internal_time):
* gst-libs/gst/audio/gstaudioclock.h:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_init), (gst_base_audio_sink_dispose),
(gst_base_audio_sink_get_time), (gst_base_audio_sink_event),
(gst_base_audio_sink_render),
(gst_base_audio_sink_create_ringbuffer),
(gst_base_audio_sink_change_state):
Make sure the audio clock always returns an increasing value.
2005-07-20 09:08:05 +00:00
Wim Taymans
c84a6b964f gst-libs/gst/audio/gstbaseaudiosink.c: Align samples even if we have roundoff errors in the timestamp conversion.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Align samples even if we have roundoff errors in the
timestamp conversion.
2005-07-16 17:13:35 +00:00
Wim Taymans
82dc411e33 Updated seek example.
Original commit message from CVS:
* docs/libs/tmpl/gstringbuffer.sgml:
* examples/seeking/seek.c: (make_vorbis_theora_pipeline),
(query_rates), (query_positions_elems), (query_positions_pads),
(update_scale), (do_seek):
Updated seek example.

* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_pad_submit_page), (gst_ogg_demux_activate_chain),
(gst_ogg_demux_find_chains), (gst_ogg_demux_send_event),
(gst_ogg_demux_loop):
Push out correct discont values.

* ext/theora/theoradec.c: (theora_dec_src_convert),
(theora_dec_sink_convert), (theora_dec_src_getcaps),
(theora_dec_sink_event), (theora_handle_type_packet),
(theora_handle_header_packet), (theora_dec_push),
(theora_handle_data_packet), (theora_dec_chain),
(theora_dec_change_state):
Better timestamping.

* ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init),
(vorbis_dec_sink_event), (vorbis_dec_push),
(vorbis_handle_data_packet), (vorbis_dec_chain):
* ext/vorbis/vorbisdec.h:
Better timestamping.

* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_time), (gst_base_audio_sink_get_times),
(gst_base_audio_sink_event), (gst_base_audio_sink_render):
Handle syncing on timestamps instead of sample offsets. Make
use of DISCONT values as described in design docs.

* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_get_time):
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_acquire),
(gst_ring_buffer_set_sample), (gst_ring_buffer_commit),
(gst_ring_buffer_read):
* gst-libs/gst/audio/gstringbuffer.h:
* sys/ximage/ximagesink.c: (gst_ximagesink_get_times),
(gst_ximagesink_show_frame):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_times):
Correcly convert buffer timestamp to stream time.
2005-07-16 14:47:27 +00:00
Thomas Vander Stichele
d143d256a6 more autistic cleanliness in functions/names/defines
Original commit message from CVS:
more autistic cleanliness in functions/names/defines
2005-07-14 09:35:11 +00:00
Thomas Vander Stichele
5852e82a04 install but don't dist the enumtypes
Original commit message from CVS:
install but don't dist the enumtypes
2005-07-13 19:28:39 +00:00
Thomas Vander Stichele
3489636a8d install the enumtypes header because audio plugins in other modules need it
Original commit message from CVS:
install the enumtypes header because audio plugins in other modules need it
2005-07-13 19:25:23 +00:00
Thomas Vander Stichele
9e8a11d3ce use overridable ERROR_CFLAGS; more macro splitting
Original commit message from CVS:
use overridable ERROR_CFLAGS; more macro splitting
2005-07-10 12:03:58 +00:00
Wim Taymans
ceb88a7777 Added audiosource base classes.
Original commit message from CVS:
Added audiosource base classes.
Ported alsasrc, still very basic.
2005-07-06 15:27:17 +00:00
Andy Wingo
e4180644b1 Many files: Null if we got it....
Original commit message from CVS:
2005-07-05  Andy Wingo  <wingo@pobox.com>

* Many files: Null if we got it....
2005-07-05 11:08:56 +00:00
Wim Taymans
2e2623748d gst-libs/gst/audio/: Fix compilation error.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_init):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_baseaudiosink_class_init), (gst_baseaudiosink_dispose),
(gst_baseaudiosink_change_state):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ringbuffer_set_callback):
Fix compilation error.
Ringbuffer starts out as not running.
Free our clock in dispose.
When releasing the ringbuffer we need to renegotiate so
clear the pad caps.
2005-06-29 11:17:33 +00:00
Thomas Vander Stichele
36d0c9ce29 reinstate plugin docs
Original commit message from CVS:
reinstate plugin docs
2005-06-29 10:56:25 +00:00
Andy Wingo
d0bc038021 *.c: Don't cast to GstObject before reffing/unreffing.
Original commit message from CVS:
2005-06-28  Andy Wingo  <wingo@pobox.com>

* *.c: Don't cast to GstObject before reffing/unreffing.
2005-06-28 10:16:13 +00:00
Jan Schmidt
2255ffd384 gst-libs/gst/audio/gstaudiosink.c: Set the worker thread's running flag to TRUE before starting the thread.
Original commit message from CVS:
2005-06-25  Jan Schmidt  <thaytan@mad.scientist.com>
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
Set the worker thread's running flag to TRUE before starting the
thread.
* gst/playback/gstdecodebin.c: (gst_decode_bin_init):
Catch a failure to add typefind to the bin.
2005-06-24 16:15:25 +00:00
Wim Taymans
6a5293d065 gst-libs/gst/audio/gstringbuffer.c: Don't try to call the delay method when the device is not opened.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_delay):
Don't try to call the delay method when the device is not
opened.
2005-05-31 11:38:10 +00:00
Wim Taymans
5474600d4f gst-libs/gst/audio/: Various small cleanups.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
(audioringbuffer_thread_func), (gst_audioringbuffer_init),
(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
(gst_audioringbuffer_play), (gst_audioringbuffer_stop),
(gst_audioringbuffer_delay), (gst_audiosink_class_init),
(gst_audiosink_create_ringbuffer):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_baseaudiosink_class_init), (gst_baseaudiosink_init),
(gst_baseaudiosink_get_clock), (gst_baseaudiosink_get_time),
(gst_baseaudiosink_set_property), (build_linear_format),
(debug_spec_caps), (debug_spec_buffer),
(gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times),
(gst_baseaudiosink_event), (gst_baseaudiosink_preroll),
(gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer),
(gst_baseaudiosink_callback), (gst_baseaudiosink_change_state):
* gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type),
(gst_ringbuffer_set_callback), (gst_ringbuffer_acquire),
(gst_ringbuffer_release), (gst_ringbuffer_is_acquired),
(gst_ringbuffer_play), (gst_ringbuffer_pause),
(gst_ringbuffer_stop), (gst_ringbuffer_delay),
(gst_ringbuffer_played_samples), (gst_ringbuffer_set_sample),
(wait_segment), (gst_ringbuffer_commit),
(gst_ringbuffer_prepare_read), (gst_ringbuffer_advance),
(gst_ringbuffer_clear):
Various small cleanups.

* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
(gst_audio_convert_change_state):
* gst/subparse/gstsubparse.c: (gst_subparse_chain):
No need to take the locks anymore.
2005-05-25 19:52:14 +00:00
Ronald S. Bultje
a159660dfc gst-libs/gst/audio/gstringbuffer.c: This can't be good.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_clear):
This can't be good.
2005-05-23 18:07:28 +00:00
Wim Taymans
9fccefe949 gst/: Fix passthrough in ffmpegcolorspace.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_link),
(gst_audiofilter_chain):
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
(audioringbuffer_thread_func), (gst_audioringbuffer_init),
(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
(gst_audioringbuffer_play), (gst_audioringbuffer_stop),
(gst_audioringbuffer_delay), (gst_audiosink_class_init),
(gst_audiosink_create_ringbuffer):
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
(gst_audio_convert_caps_remove_format_info),
(gst_audio_convert_getcaps), (gst_audio_convert_setcaps),
(gst_audio_convert_fixate), (gst_audio_convert_channels):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
(gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_chain):
Fix passthrough in ffmpegcolorspace.
Fix memset in audiosink on wrong memory.
2005-05-17 10:47:02 +00:00
David Schleef
d90ee5bfa3 Port from GstData to GstMiniObject.
Original commit message from CVS:
Port from GstData to GstMiniObject.
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
(gst_ogg_mux_queue_pads), (gst_ogg_mux_set_header_on_caps),
(gst_ogg_mux_collected):
* ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
* ext/theora/theoradec.c: (theora_handle_comment_packet),
(theora_handle_data_packet):
* ext/theora/theoraenc.c: (theora_buffer_from_packet),
(theora_set_header_on_caps), (theora_enc_chain):
* ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
(vorbis_handle_comment_packet):
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_set_header_on_caps):
* ext/vorbis/vorbisparse.c: (vorbis_parse_set_header_on_caps):
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_chain):
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_chain):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_get_buffer):
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
* gst/playback/gstplaybasebin.c: (check_queue), (probe_triggered),
(mute_stream), (silence_stream):
* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
* gst/volume/gstvolume.c: (volume_transform):
* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
(gst_ximage_buffer_init), (gst_ximage_buffer_class_init),
(gst_ximage_buffer_get_type), (gst_ximagesink_check_xshm_calls),
(gst_ximagesink_ximage_new), (gst_ximagesink_ximage_destroy),
(gst_ximagesink_ximage_put), (gst_ximagesink_imagepool_clear),
(gst_ximagesink_show_frame), (gst_ximagesink_buffer_free),
(gst_ximagesink_buffer_alloc):
* sys/ximage/ximagesink.h:
2005-05-16 15:35:52 +00:00
Jan Schmidt
8a124c2c66 configure.ac: Disable libvisual
Original commit message from CVS:
* configure.ac:
Disable libvisual

* examples/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
Fixups for missing variables.
2005-05-09 11:55:12 +00:00
Wim Taymans
fa8c2eb659 Make the base audiosink return an error when there is no audiobuffer negotiated.
Original commit message from CVS:
Make the base audiosink return an error when there is no
audiobuffer negotiated.
2005-05-06 16:18:24 +00:00
Wim Taymans
b9a30899dd GCC 4 compile fixes
Original commit message from CVS:
GCC 4 compile fixes
2005-05-05 10:42:41 +00:00
Wim Taymans
658bd2cac6 More work on the audiosink, mostly debugging and a race in shutdown.
Original commit message from CVS:
* docs/design-audiosinks.txt:
* gst-libs/gst/audio/TODO:
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
(audioringbuffer_thread_func), (gst_audioringbuffer_init),
(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
(gst_audioringbuffer_play), (gst_audioringbuffer_stop),
(gst_audioringbuffer_delay), (gst_audiosink_class_init),
(gst_audiosink_create_ringbuffer):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_baseaudiosink_class_init), (gst_baseaudiosink_init),
(gst_baseaudiosink_get_clock), (gst_baseaudiosink_get_time),
(gst_baseaudiosink_set_property), (build_linear_format),
(debug_spec_caps), (debug_spec_buffer),
(gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times),
(gst_baseaudiosink_event), (gst_baseaudiosink_preroll),
(gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer),
(gst_baseaudiosink_callback), (gst_baseaudiosink_change_state):
* gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type),
(gst_ringbuffer_set_callback), (gst_ringbuffer_acquire),
(gst_ringbuffer_release), (gst_ringbuffer_play),
(gst_ringbuffer_pause), (gst_ringbuffer_stop),
(gst_ringbuffer_delay), (gst_ringbuffer_played_samples),
(gst_ringbuffer_set_sample), (wait_segment),
(gst_ringbuffer_commit), (gst_ringbuffer_prepare_read),
(gst_ringbuffer_advance), (gst_ringbuffer_clear):
More work on the audiosink, mostly debugging and a race in
shutdown.
2005-05-05 09:37:46 +00:00
Wim Taymans
235ea5989c Make ringbuffer faster and more simple by removing the locks in the playback thread.
Original commit message from CVS:
Make ringbuffer faster and more simple by removing the locks
in the playback thread.
Add sample accurate playback based on buffer sample offsets.
Make the baseaudiosink provide a clock.
Parse caps in the base class.
Correctly handle seeking, flushing and state changes.
2005-04-28 16:15:42 +00:00
David Schleef
ab06cc8f10 Don't use GST_PLUGIN_LDFLAGS, because these aren't plugins.
Original commit message from CVS:
Don't use GST_PLUGIN_LDFLAGS, because these aren't plugins.
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
* gst-libs/gst/xwindowlistener/Makefile.am:
Convert to 0.9 API, seems to work:
* sys/ximage/Makefile.am:
* sys/ximage/ximagesink.c:
2005-04-25 07:06:09 +00:00
David Schleef
129c7e8af1 configure.ac: Remove idct and resample libs
Original commit message from CVS:
* configure.ac: Remove idct and resample libs
* gst-libs/gst/Makefile.am: same
Remove usage of gst_library_load():
* ext/alsa/gstalsaplugin.c: (plugin_init):
* ext/libvisual/visual.c: (plugin_init):
* ext/ogg/gstogg.c: (plugin_init):
* ext/theora/theora.c: (plugin_init):
* ext/vorbis/vorbis.c: (plugin_init):
* gst-libs/gst/audio/gstaudiofiltertemplate.c: (plugin_init):
* gst/audioscale/gstaudioscale.c:
* gst/adder/gstadder.c: (plugin_init):
* gst/audioconvert/plugin.c: (plugin_init):
* sys/ximage/ximagesink.c: (plugin_init):
* sys/xvimage/xvimagesink.c: (plugin_init):
* gst/tcp/gsttcpplugin.c: (plugin_init):
Link plugins against libraries:
* ext/ogg/Makefile.am:
* ext/theora/Makefile.am:
* ext/vorbis/Makefile.am:
* gst/audioconvert/Makefile.am:
Create proper libraries:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/video/Makefile.am:
Move resample library to audioscale plugin directory:
* gst-libs/gst/resample/Makefile.am:
* gst-libs/gst/resample/README:
* gst-libs/gst/resample/dtof.c:
* gst-libs/gst/resample/dtos.c:
* gst-libs/gst/resample/functable.c:
* gst-libs/gst/resample/private.h:
* gst-libs/gst/resample/resample.c:
* gst-libs/gst/resample/resample.h:
* gst-libs/gst/resample/resample.vcproj:
* gst-libs/gst/resample/test.c:
* gst/audioscale/Makefile.am:
* gst/audioscale/README:
* gst/audioscale/dtof.c:
* gst/audioscale/dtos.c:
* gst/audioscale/functable.c:
* gst/audioscale/private.h:
* gst/audioscale/resample.c:
* gst/audioscale/resample.h:
* gst/audioscale/test.c:
Move tagedit library to gst-libs:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/gstid3tag.c:
* gst-libs/gst/tag/gsttagediting.c:
* gst-libs/gst/tag/gsttageditingprivate.h:
* gst-libs/gst/tag/gstvorbistag.c:
* gst/tags/Makefile.am:
* gst/tags/gstid3tag.c:
* gst/tags/gstvorbistag.c:
Fix for core changes:
* gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
(gst_sinesrc_init), (gst_sinesrc_src_fixate), (gst_sinesrc_link),
(gst_sinesrc_getrange):
2005-04-25 00:23:06 +00:00
Wim Taymans
5a3941c762 An attempt at a set of audio base classes together with some design docs.
Original commit message from CVS:
* docs/design-audiosinks.txt:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/TODO:
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
(audioringbuffer_thread_func), (gst_audioringbuffer_init),
(gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize),
(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
(gst_audioringbuffer_play), (gst_audioringbuffer_stop),
(gst_audioringbuffer_delay), (gst_audiosink_base_init),
(gst_audiosink_class_init), (gst_audiosink_init),
(gst_audiosink_create_ringbuffer):
* gst-libs/gst/audio/gstaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init),
(gst_baseaudiosink_init), (gst_baseaudiosink_set_property),
(gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps),
(gst_baseaudiosink_get_times), (gst_baseaudiosink_event),
(gst_baseaudiosink_preroll), (gst_baseaudiosink_render),
(gst_baseaudiosink_create_ringbuffer),
(gst_baseaudiosink_callback), (gst_baseaudiosink_change_state):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type),
(gst_ringbuffer_class_init), (gst_ringbuffer_init),
(gst_ringbuffer_dispose), (gst_ringbuffer_finalize),
(gst_ringbuffer_set_callback), (gst_ringbuffer_acquire),
(gst_ringbuffer_release), (gst_ringbuffer_play_unlocked),
(gst_ringbuffer_play), (gst_ringbuffer_pause),
(gst_ringbuffer_resume), (gst_ringbuffer_stop),
(gst_ringbuffer_callback), (gst_ringbuffer_delay),
(gst_ringbuffer_played_samples), (gst_ringbuffer_commit),
(gst_ringbuffer_prepare_read), (gst_ringbuffer_clear):
* gst-libs/gst/audio/gstringbuffer.h:
An attempt at a set of audio base classes together with some
design docs.
2005-04-20 10:19:54 +00:00
Wim Taymans
73d7c02993 Make gnomevfssrc extend the source base class.
Original commit message from CVS:
* ext/gnomevfs/Makefile.am:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_get_type),
(gst_gnomevfssrc_class_init), (gst_gnomevfssrc_init),
(gst_gnomevfssrc_set_property), (gst_gnomevfssrc_get_property),
(gst_gnomevfssrc_create), (gst_gnomevfssrc_is_seekable),
(gst_gnomevfssrc_get_size), (gst_gnomevfssrc_start),
(gst_gnomevfssrc_stop):
* ext/ogg/Makefile.am:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_get_data),
(gst_ogg_demux_find_chains), (gst_ogg_demux_sink_activate):
* ext/theora/Makefile.am:
* ext/theora/theoradec.c: (_inc_granulepos),
(theora_dec_sink_event), (theora_dec_chain):
* ext/vorbis/Makefile.am:
* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
(vorbis_dec_sink_event), (vorbis_dec_chain):
* gst-libs/gst/audio/Makefile.am:
* sys/xvimage/Makefile.am:
Make gnomevfssrc extend the source base class.
Fix linking against libs in various plugins.
2005-04-06 17:33:07 +00:00
Wim Taymans
1dae961cbf Plugin port to 0.9, ogg/theora playback should work in the seek example now.
Original commit message from CVS:
Plugin port to 0.9, ogg/theora playback should work in the seek
example now.
Removed old examples.
Removed old oggvorbisenc, renamed rawvorbisenc to vorbisenc as
explained in 0.9 TODO doc.
2005-03-31 09:43:49 +00:00
Tim-Philipp Müller
38f505b3c8 Add G_BEGIN_DECLS and G_END_DECLS around headers where missing, so that they work when included from C++ code
Original commit message from CVS:
Add G_BEGIN_DECLS and G_END_DECLS around headers where missing, so that they work when included from C++ code
2005-02-09 22:31:05 +00:00
Thomas Vander Stichele
c5c4fe1d26 ignore more
Original commit message from CVS:
ignore more
2005-01-17 12:45:48 +00:00
Thomas Vander Stichele
802e07ad33 ignore generated files
Original commit message from CVS:
ignore generated files
2005-01-17 12:38:17 +00:00
Ronald S. Bultje
405f640d2a gst-libs/gst/audio/Makefile.am: Try to fix buildbot.
Original commit message from CVS:
* gst-libs/gst/audio/Makefile.am:
Try to fix buildbot.
2004-12-16 19:45:32 +00:00
Martin Soto
8ed05be101 gst-libs/gst/audio/audioclock.c (gst_audio_clock_set_active)
Original commit message from CVS:
2004-11-27  Martin Soto  <martinsoto@users.sourceforge.net>

* gst-libs/gst/audio/audioclock.c (gst_audio_clock_set_active)
(gst_audio_clock_get_internal_time):
Fix active <-> inactive transitions: ensure time value always
grows and avoid abrupt value changes.
2004-11-27 09:37:20 +00:00
Ronald S. Bultje
3a0a2898af Surround sound support.
Original commit message from CVS:
* ext/a52dec/gsta52dec.c: (gst_a52dec_channels), (gst_a52dec_push),
(gst_a52dec_reneg), (gst_a52dec_loop), (plugin_init):
* ext/alsa/gstalsa.c: (gst_alsa_get_caps):
* ext/alsa/gstalsaplugin.c: (plugin_init):
* ext/dts/gstdtsdec.c: (gst_dtsdec_channels),
(gst_dtsdec_renegotiate), (gst_dtsdec_loop), (plugin_init):
* ext/faad/gstfaad.c: (gst_faad_init), (gst_faad_chanpos_from_gst),
(gst_faad_chanpos_to_gst), (gst_faad_sinkconnect),
(gst_faad_srcgetcaps), (gst_faad_srcconnect), (gst_faad_chain),
(gst_faad_change_state), (plugin_init):
* ext/faad/gstfaad.h:
* ext/vorbis/vorbis.c: (plugin_init):
* ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/audio.c: (plugin_init):
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions),
(gst_audio_get_channel_positions),
(gst_audio_set_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(add_list_to_struct), (gst_audio_set_caps_channel_positions_list),
(gst_audio_fixate_channel_positions):
* gst-libs/gst/audio/multichannel.h:
* gst-libs/gst/audio/testchannels.c: (main):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init), (gst_audio_convert_init),
(gst_audio_convert_dispose), (gst_audio_convert_getcaps),
(gst_audio_convert_parse_caps), (gst_audio_convert_link),
(gst_audio_convert_fixate), (gst_audio_convert_channels):
* gst/audioconvert/plugin.c: (plugin_init):
Surround sound support.
2004-11-25 20:36:29 +00:00
Benjamin Otte
37af33bdda ext/alsa/gstalsa.c: buffer-frames property was missing
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_get_caps_internal):
buffer-frames property was missing
* ext/arts/gst_arts.c:
rate missing from sinkcaps
* ext/audiofile/gstafparse.c:
* ext/audiofile/gstafsink.c:
* ext/audiofile/gstafsrc.c:
* ext/swfdec/gstswfdec.c:
int audio doesn't know buffer-frames
* ext/cdparanoia/gstcdparanoia.c:
int audio doesn't know chunksize either
* ext/nas/nassink.c:
it's endianness, not endianess
* gst-libs/gst/audio/audio.h:
make float standard pad template caps really describe float
* gst/law/mulaw.c: (linear_factory):
signed only, please
* gst/mpegstream/gstdvddemux.c:
widths of 20 are not valid
2004-11-09 06:08:22 +00:00
Zaheer Abbas Merali
1dd50473f7 gst-libs/gst/audio/gstaudiofilter.c: fix build
Original commit message from CVS:
2004-10-28  Zaheer Abbas Merali  <zaheerabbas at merali dot org>

* gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_link):
fix build
2004-10-28 12:51:23 +00:00
Benjamin Otte
4f9bfe73fd gst-libs/gst/audio/gstaudiofilter.c: fix link function to always query channels and query width for floats
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_link):
fix link function to always query channels and query width for
floats
* configure.ac:
add equalizer dir
* gst/equalizer/Makefile.am:
* gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_get_type),
(gst_iir_equalizer_base_init), (gst_iir_equalizer_class_init),
(gst_iir_equalizer_init), (gst_iir_equalizer_finalize),
(arg_to_scale), (setup_filter),
(gst_iir_equalizer_compute_frequencies),
(gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property),
(gst_iir_equalizer_filter_inplace), (gst_iir_equalizer_setup),
(plugin_init):
add an equalizer
2004-10-28 02:33:47 +00:00
Steve Lhomme
7b86915b8a more working plugins
Original commit message from CVS:
more working plugins
2004-07-27 21:41:30 +00:00
Steve Lhomme
a6c379fd7a rename GStreamer-0.8.lib to libgstreamer.lib
Original commit message from CVS:
rename GStreamer-0.8.lib to libgstreamer.lib
2004-07-27 09:57:33 +00:00