Commit graph

246 commits

Author SHA1 Message Date
Wim Taymans
b2fdf703c9 gst-libs/gst/audio/gstbaseaudiosink.c: After an interrupt (PAUSED/flush) assume that the next sample should not be al...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
After an interrupt (PAUSED/flush) assume that the next sample should not
be aligned to the previous sample. Fixes #417992.
2007-05-24 16:22:23 +00:00
Wim Taymans
9b188adc27 Small cleanups.
Original commit message from CVS:
* ext/cdparanoia/gstcdparanoiasrc.c:
(gst_cd_paranoia_src_read_sector):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Small cleanups.
* ext/theora/theoradec.c: (theora_dec_sink_event):
Fix typo.
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_set_gst_timestamp):
Add some FIXME
* gst/playback/gstdecodebin.c: (queue_underrun_cb):
And some debug info when a FIXME path is hit.
2007-05-21 10:25:44 +00:00
Stefan Kost
e7c3ddf3fc gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_change_state):
Fix typo in comment.
* gst/playback/gstdecodebin.c (gst_decode_bin_class_init,
free_dynamics, pad_probe, close_pad_link, try_to_link_1,
get_our_ghost_pad, remove_element_chain, queue_underrun_cb,
close_link):
* gst/playback/gstplaybin.c (gst_play_bin_set_property,
gen_audio_element, remove_sinks, gst_play_bin_send_event_to_sink):
Remove trailing whitespaces in comments.
* gst/volume/Makefile.am:
Fix tabs.
2007-05-18 15:23:43 +00:00
Tim-Philipp Müller
9e873a3c83 When posting a warning message because samples were dropped, post something more intelligible than he default error m...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init),
(gst_base_audio_src_create):
* po/POTFILES.in:
When posting a warning message because samples were dropped, post
something more intelligible than he default error message for clock
errors which is just confusing in this context (#432984).
2007-04-25 08:54:34 +00:00
Wim Taymans
b802dea831 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to create invalid calibration parameters by making the internal time...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving):
Don't try to create invalid calibration parameters by making the
internal time go backwards, instead make external time go forward.
2007-04-05 15:44:40 +00:00
Wim Taymans
450030ebaf gst-libs/gst/audio/gstbaseaudiosink.*: Store private stuff in GstBaseAudioSinkPrivate.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
(gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
(gst_base_audio_sink_query), (gst_base_audio_sink_get_time),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_event),
(clock_convert_external), (gst_base_audio_sink_resample_slaving),
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_handle_slaving), (gst_base_audio_sink_render),
(gst_base_audio_sink_async_play):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Store private stuff in GstBaseAudioSinkPrivate.
Add configurable clock slaving modes property.
API:: GstBaseAudioSink::slave-method property
Some more latency reporting tweaks.
Added skew based clock slaving correction and make it the default until
the resampling method is more robust.
2007-03-28 14:50:47 +00:00
Sébastien Moutte
1596dd263c gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_guint64_to_gdouble for conversion.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:(gst_base_audio_sink_render):
Use gst_guint64_to_gdouble for conversion.
* win32/MANIFEST:
Add new files to the win32 MANIFEST.
* win32/common/libgstaudio.def:
* win32/common/libgstpbutils.def:
Add new exported functions.
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstdecodebin.dsp:
* win32/vs6/libgstplaybin.dsp:
Change the link to libgstpbutils.lib.
* win32/vs6/libgstdecodebin2.dsp:
Add a new project for decodebin2.
* win32/vs6/libgstpbutils.dsp:
Add a new project for pbutils.
2007-03-10 15:59:33 +00:00
Wim Taymans
a2a8b1b8ce gst-libs/gst/audio/gstbaseaudiosink.c: Fix regression that made GStreamer skip the first samples of audio.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_async_play):
Fix regression that made GStreamer skip the first samples of audio.
Fixes #414684.
2007-03-06 12:10:08 +00:00
Wim Taymans
5ee0a694a6 gst-libs/gst/audio/gstbaseaudiosink.c: base time is irrelevant here.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_async_play):
base time is irrelevant here.
2007-03-01 17:29:55 +00:00
Wim Taymans
85c7eeecc3 gst-libs/gst/audio/: Improve debugging.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
* gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
Improve debugging.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_query), (gst_base_audio_sink_event),
(gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
Improve latency and clock slaving calculations.
Improve slave clock calibration.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_commit_full):
When we are asked to render N sample to 0 bytes, return N.
2007-03-01 17:01:43 +00:00
Wim Taymans
3c94c06c5a gst-libs/gst/audio/gstaudioclock.c: Fix clock name.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
(gst_audio_clock_new):
Fix clock name.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_init), (gst_base_audio_sink_query):
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
(gst_base_audio_src_query), (gst_base_audio_src_get_offset),
(gst_base_audio_src_create):
Improve latency query code.
Use proper clock names.
2007-02-28 15:02:25 +00:00
Andy Wingo
d9b6796d91 gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
2007-02-22  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_init): Disable pull-mode activation until we
figure out how to make audio sinks go to PLAYING.
2007-02-22 11:04:10 +00:00
Tim-Philipp Müller
2f45e10c73 gst-libs/gst/audio/audio.c: Fix documentation.
Original commit message from CVS:
* gst-libs/gst/audio/audio.c:
Fix documentation.
2007-02-16 10:19:45 +00:00
Stefan Kost
b2f9c0f289 More docs coverage and some ChangeLog surgery (add missing names)
Original commit message from CVS:
* ext/cdparanoia/gstcdparanoiasrc.h:
* ext/ogg/gstoggdemux.h:
* gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
(gst_audio_frame_length), (gst_audio_duration_from_pad_buffer),
(gst_audio_is_buffer_framed), (gst_audio_structure_set_int):
* gst-libs/gst/audio/audio.h:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/interfaces/videoorientation.h:
* gst/adder/gstadder.h:
More docs coverage and some ChangeLog surgery (add missing names)
2007-02-15 15:17:23 +00:00
Wim Taymans
a43d0f57eb gst-libs/gst/audio/gstbaseaudiosink.c: Answer latency query.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_query),
(gst_base_audio_sink_render), (gst_base_audio_sink_callback),
(gst_base_audio_sink_async_play),
(gst_base_audio_sink_change_state):
Answer latency query.
Use configured latency when syncing.
Fix clock slaving.
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
(gst_base_audio_src_query), (gst_base_audio_src_change_state):
Fix possible memleak.
Implement latency query.
Small cleanups.
2007-02-15 12:06:25 +00:00
Stefan Kost
7ee1b714f0 Add crossreferences to glib/gobject/gstream docs.
Original commit message from CVS:
* configure.ac:
* docs/libs/Makefile.am:
* docs/plugins/Makefile.am:
Add crossreferences to glib/gobject/gstream docs.
* gst-libs/gst/audio/audio.h:
Source formatting.
* gst/audiotestsrc/gstaudiotestsrc.c: (plugin_init):
Add own debug category.
2007-02-12 20:42:23 +00:00
Tim-Philipp Müller
5b499dec66 gst-libs/gst/audio/gstaudiofilter.c: Clear our formats structure and free the caps contained in it when shutting down.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_class_init), (gst_audio_filter_change_state):
Clear our formats structure and free the caps contained in it when
shutting down.
2007-02-06 09:42:05 +00:00
Andy Wingo
451ff2f992 gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
2007-02-05  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_callback): Update basesink->offset so that we
pull monotonically increasing offsets instead of, um, seeking back
to 0 each time. Fixes alsasrc ! alsasink!
2007-02-05 18:39:51 +00:00
Tim-Philipp Müller
2594880e87 gst-libs/gst/audio/: Oops, forgot to commit fixed-up example.
Original commit message from CVS:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_base_init),
(gst_audio_filter_template_class_init),
(gst_audio_filter_template_init),
(gst_audio_filter_template_set_property),
(gst_audio_filter_template_get_property),
(gst_audio_filter_template_setup),
(gst_audio_filter_template_filter),
(gst_audio_filter_template_filter_inplace), (plugin_init):
Oops, forgot to commit fixed-up example.
2007-02-03 23:28:45 +00:00
Tim-Philipp Müller
b63fff63d4 Port GstAudioFilter to 0.10. This change technically breaks but seems justifiable on the grounds that the base class ...
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type),
(gst_audio_filter_class_init), (gst_audio_filter_init),
(gst_audio_filter_set_caps),
(gst_audio_filter_class_add_pad_templates):
* gst-libs/gst/audio/gstaudiofilter.h:
Port GstAudioFilter to 0.10. This change technically breaks
API and ABI (and thus also every library developer's heart),
but seems justifiable on the grounds that the base class was
completely unusable before (ie. would crash immediately when
actually used). Fixes #403963 (and eventually also #403572).
Also document all of this a bit.
2007-02-03 20:19:35 +00:00
Andy Wingo
d853b23819 gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
2007-01-12  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_fixate): Implement, stolen from baseaudiosrc.
(gst_base_audio_sink_activate_pull): Remove the handwavey nego
stuff, as the base class handles this now. Actually tell the ring
buffer to start.
(gst_base_audio_sink_callback): Cast the ring buffer correctly.
How did this work before? Maybe I'm not as awesome a programmer as
I think.

* gst-libs/gst/audio/gstbaseaudiosrc.c
(gst_base_audio_src_fixate): Rework as a basesrc vmethod instead
of a pad function.
2007-01-12 21:19:35 +00:00
Tim-Philipp Müller
ddf40c2406 gst-libs/gst/audio/mixerutils.h: Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be used when compiling...
Original commit message from CVS:
* gst-libs/gst/audio/mixerutils.h:
Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be
used when compiling with c++ compilers as well.
2007-01-12 12:47:29 +00:00
Wim Taymans
62ef7da73b Small documentation updates/fixes
Original commit message from CVS:
* ext/theora/theoradec.c:
* ext/vorbis/vorbisdec.c:
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_commit_full):
* gst-libs/gst/audio/gstringbuffer.h:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* gst-libs/gst/tag/gstvorbistag.c:
Small documentation updates/fixes
2007-01-09 11:15:57 +00:00
Andy Wingo
85aee8e273 gst-libs/gst/audio/gstbaseaudiosink.c (gst_base_audio_sink_class_init)
Original commit message from CVS:
2007-01-06  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_class_init)
(gst_base_audio_sink_init):
(gst_base_audio_sink_activate_pull): Add an activate_pull function
to baseaudiosink, and tell basesink that we can work in pull mode.
This way the ring buffer thread drives the pipeline directly, if
pull mode is possible. There is some lingering nastiness regarding
capsnego, however.
(gst_base_audio_sink_callback): Implement the callback to pull
data. This interface is a bit light, though -- it should get a
GstFlowReturn return value at least.
2007-01-06 17:28:40 +00:00
Thomas Vander Stichele
95ada43982 configure.ac: split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS so that GST_BASE_CFLAGS can go inbetwe...
Original commit message from CVS:
* configure.ac:
split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS
so that GST_BASE_CFLAGS can go inbetween them, making sure
we use uninstalled gst-libs headers
* docs/libs/Makefile.am:
* ext/alsa/Makefile.am:
* ext/cdparanoia/Makefile.am:
* ext/gnomevfs/Makefile.am:
* ext/libvisual/Makefile.am:
* ext/ogg/Makefile.am:
* ext/theora/Makefile.am:
* ext/vorbis/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/cdda/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst/adder/Makefile.am:
* gst/audioconvert/Makefile.am:
* gst/audiorate/Makefile.am:
* gst/audioresample/Makefile.am:
* gst/playback/Makefile.am:
* gst/tcp/Makefile.am:
* gst/videoscale/Makefile.am:
* gst/volume/Makefile.am:
* sys/ximage/Makefile.am:
* sys/xvimage/Makefile.am:
* tests/icles/Makefile.am:
adapt
2007-01-04 12:49:48 +00:00
Wim Taymans
0990cbf274 gst-libs/gst/audio/gstbaseaudiosink.*: Make the clock sync code more accurate wrt resampling and playback at differen...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Make the clock sync code more accurate wrt resampling and playback
at different rates.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_commit_full), (gst_ring_buffer_commit):
* gst-libs/gst/audio/gstringbuffer.h:
Use better algorithm to interpolate sample rates.
2006-11-13 17:30:17 +00:00
Tim-Philipp Müller
7298ebaa61 gst-libs/gst/audio/: Use g_strerror instead of strerror so we get UTF-8.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
* gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
Use g_strerror instead of strerror so we get UTF-8.
2006-11-06 18:24:59 +00:00
Wim Taymans
1166abbc99 gst-libs/gst/audio/gstbaseaudiosink.*: Extract rate from the NEWSEGMENT event.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Extract rate from the NEWSEGMENT event.
Use commit_full to also take rate adjustment into account when writing
samples to the ringbuffer.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_commit_full), (gst_ring_buffer_commit),
(gst_ring_buffer_read):
* gst-libs/gst/audio/gstringbuffer.h:
Added _commit_full() to also take rate into account.
Use simple interpolation algorithm to resample audio.
API: gst_ring_buffer_commit_full()
* tests/examples/seek/scrubby.c: (speed_cb), (do_seek):
* tests/examples/seek/seek.c: (segment_done):
Don't try to seek with 0.0 rate, just pause instead.
Remove bogus debug line.
2006-10-18 13:42:49 +00:00
Ville Syrjala
9b139e41fb gst-libs/gst/audio/gstbaseaudiosink.c: Don't crash when ringbuffer is not yet created.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_setcaps):
Don't crash when ringbuffer is not yet created.
Patch by: Ville Syrjala <ville dot syrjala at movial dot fi>
Fixes #361634.
* gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
* gst/playback/gststreamselector.c:
(gst_stream_selector_request_new_pad):
Activate pads befre adding them to running elements.
2006-10-13 14:15:42 +00:00
Josep Torre Valles
4de10dacb6 ext/gnomevfs/: Fix URI interface implementation return type.
Original commit message from CVS:
2006-10-10  Zaheer Abbas Merali  <zaheerabbas at merali dot org>

Patch by: Josep Torre Valles <josep@fluendo.com>

* ext/gnomevfs/gstgnomevfssink.c:
* ext/gnomevfs/gstgnomevfssrc.c:
Fix URI interface implementation return type.
* ext/pango/gsttextoverlay.c: (gst_text_overlay_set_property):
Fix what looks like a copy/paste issue when assigning values.
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_get_type):
Cast to prevent Forte warnings.
* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
Fix URI interface implementation return type.
gst_pad_query_position requires a signed integer pointer as
3rd parameter, GstClockTime is unsigned.
* gst/audioconvert/audioconvert.c:
Fix integer overflow when treated as signed.
* gst/audioresample/resample.c: (resample_add_input_data):
Cast to prevent warnings on Forte.
* gst/ffmpegcolorspace/imgconvert.c: (build_rgb_palette):
Fix integer overflow when treated as signed.
* gst/ffmpegcolorspace/imgconvert_template.h:
Fix integer overflow when treated as signed. RGBA_OUT shifts bits.
* gst/playback/gstdecodebin.c: (queue_filled_cb),
(cleanup_decodebin):
Who initialises a guint to -1!
Cast function pointers to prevent warnings on Forte.
* gst/playback/gstplaybasebin.c: (queue_deadlock_check),
(queue_threshold_reached):
Cast function pointers correctly to prevent warnings on Forte.
* gst/playback/gststreaminfo.c: (gst_stream_info_dispose):
Cast function pointers correctly to prevent warnings on Forte.
* gst/subparse/gstssaparse.c: (gst_ssa_parse_setcaps):
Obvious change to unsigned, 0xEF > max signed char.
* gst/tcp/gstmultifdsink.c: (get_buffers_max), (count_burst_unit):
GstClockTime is unsigned, initialise correctly.
* gst/tcp/gsttcp.c: (gst_tcp_socket_write):
Cast so pointer arithemetic doesn't cause warnings on Forte.
* gst/videorate/gstvideorate.c:
Use correct return value.
* tests/examples/seek/scrubby.c:
GstClockTime is unsigned, initialise correctly.
2006-10-10 12:49:03 +00:00
Tim-Philipp Müller
9e107d670a Printf format fixes.
Original commit message from CVS:
* ext/alsa/gstalsadeviceprobe.c:
(gst_alsa_device_property_probe_get_values):
* ext/alsa/gstalsasink.c: (set_hwparams):
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_elem_pad),
(gst_ogg_chain_new_stream), (gst_ogg_demux_read_chain):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers),
(gst_ogg_mux_process_best_pad):
* ext/ogg/gstoggparse.c: (gst_ogg_parse_new_stream),
(gst_ogg_parse_chain):
* ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_setup),
(gst_vorbis_enc_buffer_check_discontinuous):
* ext/vorbis/vorbisparse.c: (vorbis_parse_src_query):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_handle_track_seek):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_push_full):
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
* gst/audioresample/resample.c: (resample_input_pushthrough):
* gst/playback/gstplaybasebin.c: (queue_out_of_data):
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
(wavpack_type_find):
* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
(gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
* tests/check/elements/volume.c: (GST_START_TEST):
Printf format fixes.
2006-10-05 15:55:21 +00:00
Wim Taymans
9945d7a468 gst-libs/gst/audio/gstbaseaudiosink.c: When we have a timestamp, we can still perform clipping.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
When we have a timestamp, we can still perform clipping.
When we have no clock, we must play the sample ASAP.
2006-09-28 15:08:15 +00:00
Wim Taymans
1980f16731 gst-libs/gst/audio/gstbaseaudiosink.c: Add some more info in a WARNING.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Add some more info in a WARNING.
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Handle PAUSE in create function, use new -core addition to
wait for playing. Fixes pausing and resuming capture from an
audiosrc.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
(gst_ring_buffer_read):
Constify some more.
Caller supports interrupted reads now.
2006-09-27 13:52:14 +00:00
Wim Taymans
7367722509 Added docs for the audio libs.
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/gstaudioclock.c:
* gst-libs/gst/audio/gstaudioclock.h:
* gst-libs/gst/audio/gstaudiosink.c:
* gst-libs/gst/audio/gstaudiosink.h:
* gst-libs/gst/audio/gstaudiosrc.c:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstringbuffer.h:
Added docs for the audio libs.
2006-09-27 11:05:08 +00:00
Wim Taymans
59b7c3104f gst-libs/gst/audio/gstringbuffer.c: Early morning compilation fix.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
(gst_ring_buffer_read):
Early morning compilation fix.
2006-09-21 05:12:18 +00:00
Stefan Kost
267a068e70 ext/gnomevfs/gstgnomevfssrc.c: Add docs about icydemux usage in connection with gnomevfssrc
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c:
Add docs about icydemux usage in connection with gnomevfssrc
* ext/libvisual/visual.c:
* ext/ogg/gstoggaviparse.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c:
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
* gst-libs/gst/audio/gstaudiosink.c:
* gst-libs/gst/audio/gstaudiosrc.c:
* gst/audiorate/gstaudiorate.c:
More G_OBJECT macro fixing.
* gst/audiotestsrc/gstaudiotestsrc.h:
Fix wrong info in header due to copy & paste
2006-09-16 21:54:48 +00:00
Wim Taymans
65b1938b38 gst-libs/gst/audio/: Do the delay calculation in the source/sink base classes as this is specific for the capture/pla...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_time), (gst_base_audio_sink_callback):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_get_time), (gst_base_audio_src_fixate),
(gst_base_audio_src_get_times), (gst_base_audio_src_get_offset),
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Do the delay calculation in the source/sink base classes as this is
specific for the capture/playback mode.
Try to fixate a bit better, like round depth up to a multiple of 8
bigger than width.
Handle underruns correctly by marking DISCONT on buffers and adjusting
timestamps to handle the gap.
Set offset/offset_end correctly on buffers.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_pause),
(gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
(gst_ring_buffer_read):
Remove resync and underrun recovery from the ringbuffer.
Fix ringbuffer read code on under/overrun.
2006-09-15 14:53:44 +00:00
Wim Taymans
557b367295 configure.ac: We require 0.10.10.1 now because of _wait_preroll().
Original commit message from CVS:
* configure.ac:
We require 0.10.10.1 now because of _wait_preroll().
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Use gst_base_sink_wait_preroll().
2006-09-15 09:13:50 +00:00
Tim-Philipp Müller
ea41bfefd7 ext/alsa/: Add support for cards that (only) do more than 8 channels, like the Delta 44 (#345188).
Original commit message from CVS:
* ext/alsa/gstalsa.c: (caps_add_channel_configuration),
(gst_alsa_detect_channels):
* ext/alsa/gstalsasink.c:
Add support for cards that (only) do more than 8 channels,
like the Delta 44 (#345188).
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions):
* gst-libs/gst/audio/multichannel.h:
API: add GST_AUDIO_CHANNEL_POSITION_NONE, which stands for an
unspecified channel position and cannot be combined with any
of the other audio channel positions; adjust position layout
checks accordingly (#345188).
2006-08-03 14:16:06 +00:00
Wim Taymans
d5a10b05c2 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to align a sample to an unknown value.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Don't try to align a sample to an unknown value.
2006-07-24 16:47:10 +00:00
Wim Taymans
f3ae89426a gst-libs/gst/audio/gstbaseaudiosink.c: When the audio clock is slaved to another clock, never try to align samples bu...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_provide_clock), (gst_base_audio_sink_render):
When the audio clock is slaved to another clock, never try to align
samples but trust the rate interpolation algorithm.
2006-07-24 15:14:17 +00:00
Wim Taymans
19cd03c607 ext/alsa/gstalsasink.c: Don't try to calculate silence samples, base class does this much better now.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
Don't try to calculate silence samples, base class does this much
better now.
* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
(gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps),
(gst_ring_buffer_acquire):
Calculate silence samples correctly.
* gst-libs/gst/audio/gstringbuffer.h:
Add _CAST macro.
2006-07-24 14:34:42 +00:00
Wim Taymans
843202b51c gst-libs/gst/audio/gstaudiosink.c: Fix leak.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
(gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire),
(gst_audioringbuffer_release), (gst_audioringbuffer_stop):
Fix leak.
Avoid type casting when we can.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_dispose):
Fix mem leak.
2006-07-21 10:43:54 +00:00
Tim-Philipp Müller
a56652b204 gst-libs/gst/audio/multichannel.c: Const-ify two arrays.
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions),
(gst_audio_fixate_channel_positions):
Const-ify two arrays.
2006-07-17 13:48:10 +00:00
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