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Author SHA1 Message Date
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
Steve Lhomme
6c8a44d0c9 avoid problems with math.h, fix release dependancy 
Original commit message from CVS:
avoid problems with math.h, fix release dependancy
 2004-07-27 09:48:51 +00:00
Steve Lhomme
30712847fe add more plugins to the build add some definitions needed by plugins fixes for build problems 
Original commit message from CVS:
add more plugins to the build
add some definitions needed by plugins
fixes for build problems
 2004-07-26 22:09:59 +00:00