1420 commits

Author	SHA1	Message	Date
Mathieu Duponchelle	d1cb9c994b	video/audio: #define metadata strings. ... For instance "orientation" becomes GST_VIDEO_ORIENTATION_METADATA.	2013-09-09 15:37:02 +02:00
Sebastian Dröge	96ab6db422	audioencoder: Simplify pushing of pending events during negotiation ... And also don't send the same caps twice.	2013-08-23 19:17:16 +02:00
Sebastian Dröge	daf017ced8	audiodecoder: Fix last commit and simplify code a lot	2013-08-23 19:10:48 +02:00
Edward Hervey	f9ebfd57f8	audiodecoder: Fix previous commit ... (sorry)	2013-08-23 16:59:30 +02:00
Edward Hervey	cd3fe60c68	audiodecoder: Don't push out identical caps ... This avoids triggering plenty of extra code/methods/overhead downstream when we can just quickly check whenever we want to set caps whether they are identical or not https://bugzilla.gnome.org/show_bug.cgi?id=706600	2013-08-23 15:22:05 +02:00
Tim-Philipp Müller	6b070784c4	audio: make direct includes work again ... Not nice to break people's code if we can avoid it. Could add a warning in the next cycle, and then require single includes in the cycle after. https://bugzilla.gnome.org/show_bug.cgi?id=695889	2013-08-16 14:14:11 +01:00
Youness Alaoui	ca2a515373	audiodecoder: Clear taglist on reception of a STREAM_START event ... https://bugzilla.gnome.org/show_bug.cgi?id=705109	2013-08-12 13:02:59 +02:00
Matej Knopp	197376212c	audiodecoder: do not leak input caps ... https://bugzilla.gnome.org/show_bug.cgi?id=704926	2013-07-26 15:37:04 +01:00
Sebastian Dröge	99ef452fc4	audio/videodecoder: Rename variable in macro from dec to __dec ... Otherwise it might shadow another variable in the outside scope and cause interesting side effects.	2013-07-25 14:11:28 +02:00
Sebastian Dröge	50fd867a43	audioencoder: Don't return not-negotiated if flushing ... If the pad is flushing after a failed negotiation, return GST_FLOW_FLUSHING instead from finish_frame(). https://bugzilla.gnome.org/show_bug.cgi?id=701763	2013-06-30 18:17:42 +02:00
Mathieu Duponchelle	97e68b36c7	audiodecoder: Don't return not-negotiated if flushing ... If the pad is flushing after a failed negotiation, return GST_FLOW_FLUSHING. https://bugzilla.gnome.org/show_bug.cgi?id=701763	2013-06-25 12:51:55 -04:00
Jonas Holmberg	82e5ec553b	audioencoder: unref before memset ... Unref allocator and input_caps in encoder context before memsetting the context.	2013-06-19 13:56:28 +02:00
Ognyan Tonchev	f240d34c7e	audiobasesrc: add 2 missing gst_buffer_unmap () calls ... There are 2 missing calls to gst_buffer_unmap () in the error handling in create (). Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702467	2013-06-17 16:34:26 +02:00
Sebastian Dröge	ff5d3313d4	Release 1.1.1	2013-06-05 18:31:27 +02:00
Sebastian Dröge	c06377b385	audioencoder: Remove private copy of gst_audio_info_is_equal() ... And improve the public one a bit based on it.	2013-06-01 09:06:22 +02:00
Sebastian Dröge	5065e76b1c	audio: Add gst_audio_info_is_equal()	2013-05-30 23:56:52 +02:00
Sebastian Dröge	b8c6413a8e	audio: Always provide a buffer in gst_audio_(enc|dec)oder_allocate_output_buffer() ... We have no way of tell the caller of the exact error (e.g. if we're flushing), so will have to wait until the caller uses API that returns a GstFlowReturn, for example when pushing this buffer. https://bugzilla.gnome.org/show_bug.cgi?id=700006	2013-05-24 16:54:46 +02:00
Alexander Schrab	a049b102da	alsasrc: Make using driver timestamps possible ... https://bugzilla.gnome.org/show_bug.cgi?id=699744	2013-05-20 11:25:17 +02:00
Sebastian Dröge	be154ee9d6	audio-info: Always pass NULL as position parameter to gst_audio_info_set_format() ... https://bugzilla.gnome.org/show_bug.cgi?id=700259	2013-05-15 09:26:56 +02:00
Sebastian Dröge	b401f447d2	audio-info: For more than 64 channels don't allow a channel layout ... More than 64 channels have all channels unpositioned. https://bugzilla.gnome.org/show_bug.cgi?id=700259	2013-05-14 09:34:21 +02:00
Sebastian Dröge	351405d8a0	audio: Make sure to push pre-caps events before the caps event	2013-05-08 15:56:34 +02:00
Tim-Philipp Müller	f5c0d61be7	Update disted orc backup files ... Generated with 0.4.17 now.	2013-04-22 13:58:33 +01:00
Sebastian Dröge	d537a21075	audioencoder: Ignore caps events if the input caps did not change	2013-04-18 09:58:36 +02:00
Sebastian Dröge	d1a08af605	audiodecoder: Ignore caps events if the input caps did not change	2013-04-18 09:58:36 +02:00
Tim-Philipp Müller	e96ca66c36	docs: add some more audio macros	2013-04-17 09:26:40 +01:00
Sebastian Dröge	98f41f1c39	audioringbuffer: Also reset segbase	2013-04-15 10:13:14 +02:00
Paul HENRYS	587b2721c8	audioringbuffer: Reset segdone when releasing audioringbuffer ... https://bugzilla.gnome.org/show_bug.cgi?id=697723	2013-04-15 10:09:49 +02:00
Wim Taymans	76d71da1c4	audiodecoder: don't make negative timestamp ... Clamp timestamp interpollation to 0 to avoid going negative. This should not happen, really, but until the interpolation is improved this seems better.	2013-03-31 13:46:30 +02:00
Wim Taymans	03f658dda2	audiodecoder: forward stream-start immediately	2013-03-30 19:14:37 +01:00
Stefan Sauer	e4ee1dde02	audioencoder: api doc fixes.	2013-03-29 10:33:35 +01:00
Paul HENRYS	78a8531c75	audiobasesrc: Fix ringbuffer handling when settings caps ... ringbuffer was released after setting values to its spec field in gst_audio_base_src_setcaps(). This led to failure in case gst_audio_base_src_setcaps() is called more than one time. https://bugzilla.gnome.org/show_bug.cgi?id=696540	2013-03-25 10:16:03 +01:00
Marc Leeman	0fa50b44f0	audioringbuffer: avoid division by 0 when outputting debug info ... https://bugzilla.gnome.org/show_bug.cgi?id=695832	2013-03-15 09:06:07 +00:00
Akihiro Tsukada	a32877125f	audio: add support for AAC pass-through ... https://bugzilla.gnome.org/show_bug.cgi?id=694443	2013-02-27 00:38:05 +00:00
Stefan Sauer	b274ff7c21	audioringbuffer: log a few more details (e.g. obj-name)	2013-02-25 19:55:00 +01:00
Tim-Philipp Müller	6682215d9d	audio: fix GST_AUDIO_INFO_ENDIANNESS macro	2013-02-16 13:06:54 +00:00
Tim-Philipp Müller	664adc6e19	gst-libs: use GST_*_1_0 environment variables everywhere ... The _1_0 suffixed environment variables override the non-suffixed ones, so if we're in an environment that sets the _1_0 suffixed ones, such as jhbuild, we need to set those to make sure ours actually always get used.	2013-01-16 10:16:27 +00:00
Tim-Philipp Müller	b4def63f55	audio: don't use uninitialized variable in debug log ... https://bugzilla.gnome.org/show_bug.cgi?id=667317	2012-12-29 14:29:53 +00:00
Wim Taymans	fe93457191	audioclock: mark as using some other clock ... We need to mark our clock as using some other clock source. Alsa source uses the clock type to decide if it can use alsa driver timestamps or not. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=690465	2012-12-20 16:48:04 +01:00
Wim Taymans	5e04fcd2ef	audiobasesrc: init variable ... We need to initialize this variable because we can't be sure that the subclass will set it.	2012-12-20 16:47:56 +01:00
Tim-Philipp Müller	68f366a8d3	audiobasesrc: bail out if subclass posts an error ... Use new ringbuffer ERROR state to make all the various threads bail out correctly when the subclass posts an error. It's a bit iffy to communicate this properly between the different bits of code. https://bugzilla.gnome.org/show_bug.cgi?id=690197	2012-12-17 20:50:32 +00:00
Tim-Philipp Müller	4f49c7a33b	audioringbuffer: add GST_AUDIO_RING_BUFFER_STATE_ERROR state ... API: GST_AUDIO_RING_BUFFER_STATE_ERROR https://bugzilla.gnome.org/show_bug.cgi?id=690197	2012-12-17 20:50:32 +00:00
Thiago Santos	929edc2572	audiobasesrc: Always resync the ringbuffer on the first buffer ... In SKEW mode, use next_sample == -1 to check for the first sample when starting to read samples so it resyncs the ringbuffer and timestamps are ok. Suggestion from Teemu Katajisto <teemu.katajisto@digia.com> https://bugzilla.gnome.org/show_bug.cgi?id=648359	2012-12-17 11:47:34 +01:00
Sebastian Dröge	3f82e919dd	libs: Use foo/foo.h as single-include header consistently everywhere ... https://bugzilla.gnome.org/show_bug.cgi?id=688785	2012-12-12 17:13:10 +00:00
Tim-Philipp Müller	fbff6c6fb1	audioencoder: add some more debug info and remove obsolete comment	2012-12-02 12:33:43 +00:00
Tim-Philipp Müller	8827437b61	audio: remove bogus Since marker from docs ... It was causing perl warnings in gtk-doc code.	2012-11-21 23:19:14 +00:00
Evan Nemerson	4d77fba46c	libs: Add missing single include headers and use them in GIRs	2012-11-21 11:01:24 +01:00
Tim-Philipp Müller	71e46b2478	gst_adapter_prev_timestamp -> gst_adapter_prev_pts ... https://bugzilla.gnome.org/show_bug.cgi?id=675598	2012-11-14 00:03:15 +00:00
Sebastian Dröge	32139f9a3d	audio: Use new GType for GThread instead of just G_TYPE_POINTER	2012-11-12 11:45:47 +01:00
Sebastian Dröge	d209727644	audiodecoder: Reset error count to 0 after successfully decoding a frame	2012-11-09 16:48:54 +01:00
Tim-Philipp Müller	5f59b4f7ee	Fix FSF address ... https://bugzilla.gnome.org/show_bug.cgi?id=687520	2012-11-03 23:05:09 +00:00
Miguel Angel Cabrera Moya	244fdcc69a	audiobasesink: use the same type as the internal type to return it ... https://bugzilla.gnome.org/show_bug.cgi?id=687466	2012-11-02 19:52:38 +00:00
Wim Taymans	5f44303925	audioringbuffer: reset spec on _release ... Reset the caps and the audioinfo when releasing the ringbuffer. Fixed a bug with reusing pulsesink.	2012-10-30 10:33:04 +00:00
Tim-Philipp Müller	a4f2df6341	Revert "g-i: change g-ir-scanner arg --library=libgstfoo-X.la to --library=gstfoo-X" ... This reverts commit e39fbe6b7e. Looks like we need to pass the full .la file after all in a setup with libtool, or it might not find the library, e.g. like ERROR: can't resolve libraries to shared libraries: gstfft-1.0 Conflicts: gst-libs/gst/audio/Makefile.am gst-libs/gst/pbutils/Makefile.am Also see https://bugzilla.gnome.org/show_bug.cgi?id=603710	2012-10-29 12:47:05 +00:00
Tim-Philipp Müller	973f4f09ea	audio: try harder to make g-i use the build-tree libgsttag ... without adding additional --library= tags, which shouldn't be there. https://bugzilla.gnome.org/show_bug.cgi?id=679315	2012-10-28 17:59:27 +00:00
Tim-Philipp Müller	e39fbe6b7e	g-i: change g-ir-scanner arg --library=libgstfoo-X.la to --library=gstfoo-X ... As it should be according to the man page. https://bugzilla.gnome.org/show_bug.cgi?id=679315	2012-10-28 17:35:57 +00:00
Mark Nauwelaerts	45d802b63f	audiodecoder: track forced decoding state	2012-10-24 14:46:22 +02:00
Sebastian Dröge	1813701ef2	audiobasesink: Add explanation to the GAP event handling code	2012-10-24 11:22:29 +02:00
Sebastian Dröge	b793d0bfae	audiobasesink: Properly handle GAP events ... These are now converted into silence buffers if they have a duration or cause the ringbuffer and clock to be started if they don't have a duration. Fixes bug #685273.	2012-10-24 11:19:05 +02:00
Tim-Philipp Müller	277ca04976	audiodecoder: don't leak message strings when error is not fatal ... https://bugzilla.gnome.org/show_bug.cgi?id=681192	2012-10-20 11:38:10 +01:00
Tim-Philipp Müller	3ee2ad255b	audiocdsrc: mention TOCs in docs	2012-10-17 19:59:57 +01:00
Mark Nauwelaerts	162433795a	audio: properly handle clipping of empty buffer	2012-10-15 18:48:01 +02:00
Josep Torra	d8d9f0db97	audiodecoder: set of base_ts for segment formats other than time ... Fixes setting of converted segment start as base_ts when estimate rate is allowed.	2012-10-11 13:17:01 +02:00
Sebastian Dröge	e779002cfd	audiodecoder: Don't unref caps twice ... Thanks to Josep Torra for noticing.	2012-10-10 15:50:31 +02:00
Wim Taymans	3591df23b1	docs: playbin2 -> playbin	2012-10-09 12:20:10 +02:00
Andoni Morales Alastruey	8a5cf5ef4d	audio/video: update documentation for vfunc's that require chaining up	2012-10-08 13:04:02 +02:00
Tim-Philipp Müller	cdb22274e6	audioencoder: make stop() vfunc also optional ... Just change default value, since we also don't want to fail if we want to deactivate and aren't active or want to activate and are already active. https://bugzilla.gnome.org/show_bug.cgi?id=685490	2012-10-04 13:40:32 +01:00
Andoni Morales Alastruey	795d366a0c	audioencoder: don't fail if the start vfunc is not implemented ... Fix behaviour to match documentation and decoder class behaviour. https://bugzilla.gnome.org/show_bug.cgi?id=685490	2012-10-04 13:14:10 +01:00
Michael Smith	92560517e8	Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base	2012-10-03 10:45:26 -07:00
Michael Smith	a29c4f9489	meta registration: use g_once functions to register these threadsafely.	2012-10-03 10:44:59 -07:00
Arun Raghavan	9f9718715a	audio: Explicitly specify endianness for IEC 61937 payloading ... This is required since some systems (DirectSound and OS X) manage the final byte order themselves. https://bugzilla.gnome.org/show_bug.cgi?id=678021	2012-09-19 09:15:16 +05:30
Mark Nauwelaerts	c629a44162	replace gst_tag_list_free with gst_tag_list_unref	2012-09-14 17:53:21 +02:00
Wim Taymans	a57198a0ba	audio: improve property description ... Improve the description of the latency-time and buffer-time properties in the audio sink and source.	2012-09-14 16:08:50 +02:00
Sebastian Dröge	6e33f2d464	audiodecoder: Don't output an (unreffed) buffer in error cases	2012-09-14 14:54:22 +02:00
Tim-Philipp Müller	f7c6aa5abd	Release 0.11.94	2012-09-14 02:47:54 +01:00
Olivier Crête	b35bc51ed6	audio: Fix annotations	2012-09-13 17:11:56 -04:00
Wim Taymans	0ce33461c8	audiosrc: check for flushing state in provide_clock ... Only provide a clock when we are not flushing, this means that we have posted a PROVIDE_CLOCK message. We used to check if we were acquired but that doesn't work anymore now that we do the negotiation async in the streaming thread: it's possible that we are still negotiating when the pipeline asks us for a clock.	2012-09-10 12:19:22 +02:00
Wim Taymans	44dab50b7a	ringbuffer: add method to check the flushing state	2012-09-10 12:19:22 +02:00
Mark Nauwelaerts	75fe950c33	gst-libs: restore original full padding	2012-09-10 11:45:44 +02:00
Pontus Oldberg	a2f8ec4f5a	ringbuffer: add support for timestamps ... Make it possible for subclasses to provide the timestamp (as an absolute time against the pipeline clock) of the last read data. Fix up alsa to provide the timestamp received from alsa. Because the alsa timestamps are in monotonic time, we can only do this when the monotonic clock has been selected as the pipeline clock. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=635256	2012-09-10 11:34:14 +02:00
Mark Nauwelaerts	a29fab200c	audio{de,en}coder: use GstClockTime parameters where appropriate ... Fixes https://bugzilla.gnome.org/show_bug.cgi?id=683672	2012-09-10 11:20:50 +02:00
Thibault Saunier	dc5bb008a3	audio: port to the new GLib thread API	2012-09-09 20:41:06 -03:00
Tim-Philipp Müller	2079a8c12b	Remove glib-compat-private.h stuff we don't need any more ... It's all been ported to the latest GLib API now.	2012-09-09 18:36:49 +01:00
Mark Nauwelaerts	c9d3f32cc9	audioencoder: plug some leaks	2012-09-06 12:16:59 +02:00
Wim Taymans	668ce33384	update for basesink change	2012-09-04 12:18:11 +02:00
Tim-Philipp Müller	a99a1042b9	gst_message_new_duration() -> gst_message_new_duration_changed()	2012-09-02 01:27:17 +01:00
Jan Schmidt	5dafecad31	audiodecoder: Handle GAP events in place of segment updates ... Use them to trigger generation of an empty output buffer or to send pending events downstream and trigger pre-roll	2012-08-31 12:42:12 -07:00
Edward Hervey	def07410ef	audiobasesink: Avoid resetting ringbuffer when not needed ... If the ringbuffer was configured to the same caps as previously, we don't need to reconfigure it.	2012-08-14 18:56:00 +02:00
Víctor Manuel Jáquez Leal	f7f0c55e5f	audiodecoder: getter for allocator ... Sometimes the decoder would use the allocator for something else than just allocating output buffers, for example, querying different parameters. This patch expose a getter accessor for the negotiated memory allocator.	2012-08-14 15:47:34 +02:00
Víctor Manuel Jáquez Leal	936ec3eb8f	audioencoder: getter for allocator ... Sometimes the encoder would use the allocator for something else than just allocating output buffers, for example, querying different parameters. This patch expose a getter accessor for the negotiated memory allocator.	2012-08-14 15:47:29 +02:00
Tim-Philipp Müller	2ff4d2efe3	audioencoder: return TRUE from _set_output_format() if all is good ... Fixes not-negotiated errors in wavpackenc unit test.	2012-08-13 23:34:52 +01:00
Sebastian Dröge	62ec7f837d	audioencoder: Let global tag events be handled the same way as other events	2012-08-09 17:06:31 +02:00
Sebastian Dröge	e9fbba63b5	audiodecoder: Let global tag events be handled the same way as other events	2012-08-09 16:55:19 +02:00
Sebastian Dröge	2a1f8a4da3	audio: Merge upstream stream tags	2012-08-09 16:24:47 +02:00
Sebastian Dröge	7f0e65bb46	audio: Always keep a complete taglist around ... Otherwise updates to the tags will cause non-updated tags to be lost downstream.	2012-08-09 15:48:03 +02:00
Sebastian Dröge	bc4d923982	audioencoder: Add negotiate vfunc that is used to negotiate with downstream ... The default implementation negotiates a buffer pool and allocator with downstream.	2012-08-09 15:27:33 +02:00
Sebastian Dröge	9309272309	audioencoder: Decouple setting of output format and downstream negotiation ... This makes the audio encoder base class more similar to the video encoder base class.	2012-08-09 15:21:01 +02:00
Sebastian Dröge	513d4f7cd1	audiodecoder: Add negotiate vfunc that is used to negotiate with downstream ... The default implementation negotiates a buffer pool and allocator with downstream.	2012-08-09 15:10:05 +02:00
Sebastian Dröge	e1702d62a0	audiodecoder: Decouple setting of output format and downstream negotiation ... This makes the audio decoder base class more similar to the video decoder base class.	2012-08-09 15:02:27 +02:00
Tim-Philipp Müller	6422f2d085	Update .gitignore	2012-08-08 09:06:30 +01:00
Tim-Philipp Müller	ca31913c04	audiocdsrc: update for TOC API change	2012-07-28 11:13:12 +01:00
Sebastian Dröge	99d73c94e9	tag: Update for taglist/tag event API changes	2012-07-28 00:35:02 +02:00
Wim Taymans	683a38ad65	update for new variable names	2012-07-27 15:24:43 +02:00
Wim Taymans	40a0624e99	audio-format: fix shift for 18 bits samples ... The 18bits of the sample are in the LSB so we need to shift them 14 positions to bring them to 32 bits.	2012-07-26 15:42:38 +02:00
Mark Nauwelaerts	c91615bd82	audio{de,en}coder: delay input caps processing until processing data ... Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680614	2012-07-26 14:35:30 +02:00
Mark Nauwelaerts	28537dc73c	audioencoder: avoid setting output caps twice ... ... which may not be handled or appreciated well downstream, e.g. muxers only performing header setup once.	2012-07-25 15:58:19 +02:00
Mark Nauwelaerts	1f962bc108	audioencoder: also consider filter caps in getcaps	2012-07-25 15:58:19 +02:00
Mark Nauwelaerts	26d74941fb	Revert "audioencoder: plug caps ref leak" ... This reverts commit 08ff5899a7. Was not a leak to begin with as we did not have ownership of caps.	2012-07-25 12:30:54 +02:00
Mark Nauwelaerts	08ff5899a7	audioencoder: plug caps ref leak	2012-07-25 11:58:26 +02:00
Mark Nauwelaerts	473371f943	audiodecoder: hold caps ref while needed	2012-07-25 11:58:26 +02:00
Mark Nauwelaerts	d55529621c	audioencoder: correctly compare audio info positions ... Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680553	2012-07-25 11:58:26 +02:00
Mark Nauwelaerts	65ea6dee60	audiodecoder: only arrange to reconfigure if data provided ... ... otherwise audio format need not be known already.	2012-07-24 14:48:59 +02:00
Mark Nauwelaerts	d63a4024b8	audiodecoder: minor doc fix	2012-07-24 12:30:21 +02:00
Wim Taymans	5ff002b47a	audio: prefix orc_* functions with audio_orc_* ... To avoid potential conflicts in other modules when statically linking	2012-07-23 17:16:34 +02:00
Sebastian Dröge	d55d7fdc38	audio: Renegotiate if necessary ... And also correct usage of the base class stream lock.	2012-07-23 12:01:12 +02:00
Sebastian Dröge	7b06c34868	audiodecoder: Handle allocation query	2012-07-23 11:42:22 +02:00
Sebastian Dröge	0814d38e98	audiodecoder: Add propose_allocation, decide_allocation vfuncs and functions to allocate buffers with information from the allocation query results	2012-07-23 10:28:05 +02:00
Sebastian Dröge	0513d3d9f4	audioencoder: Add propose_allocation, decide_allocation vfuncs and functions to allocate buffers with information from the allocation query results	2012-07-23 10:20:05 +02:00
Edward Hervey	55f692eff6	audiodecoder: Don't assert on pad caps not being set ... The decoder might have been de-activated in the meantime (resulting in NULL pad caps). If the decoder really isn't configured, then it will error out further down when checking whether the GST_AUDIO_INFO_IS_VALID() https://bugzilla.gnome.org/show_bug.cgi?id=667562	2012-07-19 10:55:53 +02:00
Evan Nemerson	7a7374f2ef	audiometa: add missing array array annotations	2012-07-17 11:07:18 +02:00
Evan Nemerson	17815020fd	audio: add missing array and element-type annotations for binary data	2012-07-17 11:06:57 +02:00
Evan Nemerson	fd91104636	audio-channels: add missing array-related annotations	2012-07-17 11:06:47 +02:00
Evan Nemerson	1606028c08	audioencoder: add missing element-type to set_headers method	2012-07-17 11:06:22 +02:00
Edward Hervey	2817bdadc9	libs: Remove "Since" markers and minor doc fixups	2012-07-13 12:11:06 +02:00
Edward Hervey	c9428c96b1	baseaudiosink: Resync when ringbuffer resets ... When the ringbuffer gets restarted (like in setcaps), we *will* have to resync against the new values. Without this we end up blindly assuming the new samples align to the old ones.	2012-07-12 09:51:35 +02:00
Sebastian Dröge	9de1b170b3	audiocdsrc: Remove the TOC query handling	2012-07-05 12:35:35 +02:00
Sebastian Dröge	0ac1596d8d	audiocdsrc: Update for TOC API changes	2012-07-05 12:29:00 +02:00
Sebastian Dröge	b362ec3a57	audiocdsrc: Only push TOC event, the TOC message is handled by the sinks	2012-07-03 17:31:54 +02:00
Tim-Philipp Müller	df70b2d2ce	audiocdsrc: send TOC event downstream if we're in continuous mode ... If we're in continuous mode where we'll play the entire CD from start to finish, send a TOC event downstream so any downstream muxers can write a TOC to indicate where the various tracks start and end.	2012-06-28 23:41:16 +01:00
Tim-Philipp Müller	b27c649a48	audiocdsrc: post TOC message on the bus on start-up ... First attempt at implement the various GstToc API bits in GstAudioCdSrc. https://bugzilla.gnome.org/show_bug.cgi?id=668996	2012-06-26 19:53:35 +01:00
Tim-Philipp Müller	a821d428bb	audio: make sure g-i doesn't parse orc-generated gstaudiopack.h file	2012-06-24 00:28:40 +01:00
Wim Taymans	c003efcc63	audiobasesink: fix for basesink API change	2012-06-18 11:40:36 +02:00
Jan Schmidt	d9740bf9ba	audio decoder: Add some debug output for bad caps from children	2012-06-12 23:52:35 +10:00
Vincent Penquerc'h	f8b8711081	audiodecoder: push queued events only when we have a first buffer ... https://bugzilla.gnome.org/show_bug.cgi?id=675812	2012-06-11 11:29:13 +01:00
Wim Taymans	9d6967fe9a	Add generated orc files	2012-06-08 17:57:43 +02:00
Wim Taymans	12ac9f0aa2	Also build the orc generated code	2012-06-08 17:57:43 +02:00
Wim Taymans	3f8c5ea036	audio: add orc enabled pack and unpack functions	2012-06-08 17:57:43 +02:00
Wim Taymans	8e393d898a	audio: add flag to mark possible unpack formats ... Make a new flag to mark formats that can be used in pack and unpack functions. Mark S32NE and F64NE as those unpack formats	2012-06-08 17:57:43 +02:00
Sebastian Dröge	462c4cc3d8	audio: Remove unused, generated marshallers	2012-06-08 11:28:56 +02:00
Wim Taymans	3da0b71876	audio: split audio header into logical parts	2012-06-08 10:10:08 +02:00
Wim Taymans	a2172bdb4b	update for tag event change	2012-06-06 13:05:47 +02:00
Sebastian Dröge	2667d4bb82	Revert "audiodecoder: Error out earlier in a few places if something goes wrong" ... This reverts commit eb68a2d5a7. This sometimes errors out too early now, needs some more thoughts.	2012-06-04 10:01:42 +02:00
Sebastian Dröge	f609b3a627	audiodecoder: Return setcaps return value instead of always TRUE	2012-06-04 09:56:30 +02:00
Sebastian Dröge	eb68a2d5a7	audiodecoder: Error out earlier in a few places if something goes wrong	2012-06-02 17:16:13 +02:00
Wim Taymans	c66da2c74b	audio: add flags for the pack/unpack functions ... Add a flag argument to the pack and unpack function so that we can expand it later when needed. We could for example prefer a High Quality pack/unpack operation later.	2012-05-29 09:54:43 +02:00
Arun Raghavan	9c29cd70ee	audio: Fix DTS IEC61937 payloading ... DTS type I-III specify the burst length in bits. Only type IV (which we do not currently support) needs it to be specified in bytes. Thanks to Julien Moutte for pointing this out.	2012-05-25 12:38:32 +02:00
Sebastian Rasmussen	b7b123964b	gst-libs: make pkg-config get path to pkg-config dirs from configure ... When --with-pkg-config-path is supplied to configure this path is now explicitly propagated to pkg-config. https://bugzilla.gnome.org/show_bug.cgi?id=673377	2012-05-05 23:26:20 +01:00
Sebastian Dröge	69b18ab09d	gst-libs: Remove interfaces libs and mixer/tuner interfaces ... The navigation interface is now in the video library.	2012-04-13 13:14:13 +02:00
Alban Browaeys	6c8abf24cf	libs: Link against internal tag library	2012-04-11 09:58:49 +02:00
Sebastian Dröge	8091546694	audio: Remove obsolete FIXME 0.11	2012-04-11 09:57:35 +02:00
Alessandro Decina	ebf80977c4	audiodecoder: don't discard timestamps when consecutive input buffers have the same ts ... Avoid pushing out buffers with the same timestamp only if the out buffers are decoded from the same input buffer. Instead keep the timestamps when upstream pushes consecutive buffers with the same ts.	2012-04-05 10:19:46 +02:00
Mark Nauwelaerts	6eeca397fc	audioencoder: plug a definite and rare leak	2012-04-04 19:57:35 +02:00
Sebastian Dröge	65307dd132	gst: Update versioning	2012-04-04 14:55:15 +02:00
Mark Nauwelaerts	91aa1eb7dd	audio{de,en}coder: fixup documentation	2012-04-02 14:23:33 +02:00
Sebastian Dröge	b701534204	audioencoder: Fix handling of offset/offset-end for Ogg codecs ... Fixes the vorbisenc unit test.	2012-03-31 12:55:15 +02:00
Sebastian Dröge	a103fa85a9	audio{en,de}coder: Track input and output segments separately ... They can go out of sync for some time if processing of buffers on the old segment happens after the segment was received.	2012-03-30 13:21:09 +02:00
Sebastian Dröge	9cd9f00799	audioencoder: Add gst_audio_encoder_set_headers() to the docs	2012-03-30 12:57:02 +02:00
Sebastian Dröge	78bcb67ea5	audioencoder: Add function to set in-stream headers ... API: gst_audio_encoder_set_headers() This makes the hack in vorbisenc and probably others in ::pre_push() unnecessary.	2012-03-30 12:47:28 +02:00
Sebastian Dröge	f791ec1f10	audioencoder: Rename ::event() to ::sink_event() and add ::src_event()	2012-03-30 12:23:13 +02:00
Sebastian Dröge	d8cb235fe4	audiodecoder: Rename ::event() to ::sink_event() and add ::src_event()	2012-03-30 12:23:13 +02:00
Sebastian Dröge	40a4f2f8aa	audiodecoder: Rename _byte_time() to _estimate_rate() ... Which is telling more about what this actually does and is more consistent with the video base classes.	2012-03-30 11:51:47 +02:00
Mark Nauwelaerts	2ddc6bb63d	audiodecoder: handle downstream seeking query ... ... or not, in line with how segment events are treated.	2012-03-28 16:41:01 +02:00
Wim Taymans	77a4f5865b	audioencoder: avoid caps copy	2012-03-27 15:44:43 +02:00
Wim Taymans	32bd12dba9	Merge branch 'master' into 0.11 ... Conflicts: .gitignore common configure.ac ext/vorbis/gstvorbisdeclib.h gst-libs/gst/audio/gstaudioencoder.c gst-libs/gst/riff/riff-read.c gst/playback/gstplaysink.c gst/playback/gstplaysinkconvertbin.c tests/check/libs/video.c	2012-03-22 11:35:13 +01:00
Wim Taymans	a619d3a8b0	update for memory api changes	2012-03-20 13:20:36 +01:00
Mark Nauwelaerts	278b0f093b	audio: include audio enumtypes	2012-03-19 16:18:56 +01:00
Wim Taymans	dfb8e7cb2c	don't pass random pointers to pull_range	2012-03-16 21:46:47 +01:00
Wim Taymans	4e1ed6f649	audio: fix debug line	2012-03-13 12:39:52 +01:00
Wim Taymans	25137962ad	fix for caps API changes	2012-03-11 19:04:41 +01:00
Wim Taymans	7296ef7c63	audiobasesink: add some G_LIKELY	2012-03-09 17:15:38 +01:00
Wim Taymans	94869bff38	audio: avoid buffer copy when nothing is clipped ... when nothing is clipped, return the input buffer instead of creating and returning an identical copy.	2012-03-09 16:17:54 +01:00
Sebastian Dröge	7ff608889a	audio{en,de}coder: Add optional open/close vfuncs ... This can be used to do something in NULL->READY, like checking if a hardware codec is actually available and to error out early.	2012-03-09 10:56:07 +01:00
Tim-Philipp Müller	29c266ccff	Merge remote-tracking branch 'origin/master' into 0.11 ... Conflicts: common docs/libs/gst-plugins-base-libs.types ext/pango/gsttextoverlay.c ext/vorbis/gstvorbisdec.c gst/playback/gstplaysink.c gst/playback/gstplaysinkconvertbin.c sys/ximage/ximagesink.c sys/xvimage/xvimagesink.c	2012-03-08 20:31:34 +00:00
Mark Nauwelaerts	8a3f818dce	audiodecoder: add some tag handling convenience help	2012-03-06 16:17:37 +01:00
Mark Nauwelaerts	5a0fff76f3	audiodecoder: add baseclass _CAST macro	2012-03-06 16:17:33 +01:00
Mark Nauwelaerts	d19f5467cc	audio: add helper function to convert mask to channel positions ... ... as there may be other than raw audio formats using a channel mask, and there is already one to convert the other way around.	2012-03-05 13:03:57 +01:00
Mark Nauwelaerts	debbc75272	audioencoder: stop proxying some old-style 0.10 raw audio caps fields	2012-03-05 13:03:57 +01:00
Mark Nauwelaerts	1a2863bf33	audioencoder: store segment event as pending event to forego dropping it	2012-03-05 13:03:57 +01:00
Mark Nauwelaerts	aae64c40a8	audiodecoder: plug caps leak when setting output format	2012-03-05 13:03:57 +01:00
Mark Nauwelaerts	3b0a2a60da	audiodecoder: enhance some debug statement	2012-03-05 11:04:20 +01:00
Sebastian Dröge	f7939bb43f	Merge branch 'master' into 0.11 ... Conflicts: NEWS RELEASE configure.ac docs/plugins/gst-plugins-base-plugins.args docs/plugins/gst-plugins-base-plugins.hierarchy docs/plugins/gst-plugins-base-plugins.interfaces docs/plugins/inspect/plugin-adder.xml docs/plugins/inspect/plugin-alsa.xml docs/plugins/inspect/plugin-app.xml docs/plugins/inspect/plugin-audioconvert.xml docs/plugins/inspect/plugin-audiorate.xml docs/plugins/inspect/plugin-audioresample.xml docs/plugins/inspect/plugin-audiotestsrc.xml docs/plugins/inspect/plugin-cdparanoia.xml docs/plugins/inspect/plugin-encoding.xml docs/plugins/inspect/plugin-ffmpegcolorspace.xml docs/plugins/inspect/plugin-gdp.xml docs/plugins/inspect/plugin-gio.xml docs/plugins/inspect/plugin-gnomevfs.xml docs/plugins/inspect/plugin-libvisual.xml docs/plugins/inspect/plugin-ogg.xml docs/plugins/inspect/plugin-pango.xml docs/plugins/inspect/plugin-playback.xml docs/plugins/inspect/plugin-subparse.xml docs/plugins/inspect/plugin-tcp.xml docs/plugins/inspect/plugin-theora.xml docs/plugins/inspect/plugin-typefindfunctions.xml docs/plugins/inspect/plugin-uridecodebin.xml docs/plugins/inspect/plugin-videorate.xml docs/plugins/inspect/plugin-videoscale.xml docs/plugins/inspect/plugin-videotestsrc.xml docs/plugins/inspect/plugin-volume.xml docs/plugins/inspect/plugin-vorbis.xml docs/plugins/inspect/plugin-ximagesink.xml docs/plugins/inspect/plugin-xvimagesink.xml gst-libs/gst/app/gstappsink.c gst-libs/gst/audio/mixer.c gst-libs/gst/audio/mixer.h gst-libs/gst/tag/gstxmptag.c gst-libs/gst/video/colorbalance.c gst-libs/gst/video/colorbalance.h gst/adder/gstadder.c gst/playback/gstplaybasebin.c gst/playback/gstplaybin2.c gst/playback/gstplaysink.c gst/videoscale/gstvideoscale.c tests/check/elements/videoscale.c tests/examples/seek/seek.c tests/examples/v4l/probe.c win32/common/_stdint.h win32/common/audio-enumtypes.c win32/common/config.h	2012-03-02 10:00:55 +01:00
Wim Taymans	502c12f827	update for metadata API changes	2012-02-29 17:25:10 +01:00
Wim Taymans	a232714065	meta: add return value to transform	2012-02-28 16:18:30 +01:00
Wim Taymans	1c05eeece5	update for metadata tags	2012-02-28 12:10:14 +01:00
Philippe Normand	63ace8872d	audio: link against libm ... It is used in gststreamvolume.	2012-02-27 14:36:25 +00:00
Edward Hervey	59918e841f	Suppress deprecation warnings in selected files, for g_value_array_* mostly	2012-02-27 14:28:15 +01:00
Wim Taymans	5a0354b416	audioencoder: don't leak event	2012-02-27 13:08:36 +01:00
Wim Taymans	15eb385412	audioencoder: use default event function ... Implement a default event function so that subclasses can call it without having to return FALSE (and make it impossible to report errors).	2012-02-27 12:49:52 +01:00
Wim Taymans	525f330142	update for metadata changes	2012-02-24 10:26:04 +01:00
Wim Taymans	268d52fd33	Merge branch 'master' into 0.11 ... Conflicts: gst-libs/gst/rtsp/gstrtspconnection.c win32/common/libgstaudio.def	2012-02-17 23:46:17 +01:00
Tim-Philipp Müller	0f6c8a27a7	docs: add new audio base class API to docs and .def file	2012-02-17 15:08:36 +00:00
Wim Taymans	e44dd9db8f	Merge branch 'master' into 0.11 ... Conflicts: gst-libs/gst/audio/gstaudioencoder.c gst-libs/gst/pbutils/gstdiscoverer.c	2012-02-16 14:23:28 +01:00
Mark Nauwelaerts	439884d628	audiodecoder: add some properties to tweak baseclass behaviour ... ... so subclass can also rely upon never being bothered with some NULL buffer it can't do any interesting with, or with any data before it received any format configuration (and setup properly).	2012-02-16 12:35:53 +01:00
Mark Nauwelaerts	5b4dc02523	audioencoder: add some properties to tweak baseclass behaviour ... ... so subclass can also rely upon never being bothered with less data than it desires or with some NULL buffer it can't do any interesting with.	2012-02-16 12:35:51 +01:00
Mark Nauwelaerts	95306e8fef	audiodecoder: assert some more that subclass parsed frame has proper len	2012-02-16 12:35:40 +01:00
Wim Taymans	c7d0fb556f	audiodecoder: chain up to parent for defaults ... Chain up to the parent instead of using the FALSE return value from the event function (because it's otherwise impossible to return an error).	2012-02-15 13:42:19 +01:00
Wim Taymans	b2fbb2e587	audiodecoder: call default event handler ... Call the default event handler for unknown events.	2012-02-15 13:03:59 +01:00
Wim Taymans	a75e9102c5	GST_FLOW_WRONG_STATE -> GST_FLOW_FLUSHING	2012-02-08 15:17:49 +01:00
Mark Nauwelaerts	97d60612a4	audiodecoder: remove stray obsolete declaration	2012-02-06 22:10:28 +01:00
Mark Nauwelaerts	2bf1a4428e	audio: correctly fill in fallback channel positions in stereo case	2012-02-06 22:10:28 +01:00
Wim Taymans	6c08f53416	audiofilter: configure info after calling vmethod ... First call the vmethod and then configure the audioinfo in the baseclass. This allows subclasses to know about the old format.	2012-02-06 13:23:26 +01:00
Wim Taymans	fe3e9b90dd	audioencoder: don't unref caps parameter ... Fix refcounting on incomming caps to make sure we don't unref it too much.	2012-02-03 09:51:00 +01:00
Sebastian Dröge	1cb4029d00	audioencoder: gst_pad_get_pad_template_caps() now returns a new reference, don't forget to unref	2012-02-01 16:33:30 +01:00
Sebastian Dröge	5aa6748151	audio{enc,dec}oder: Check if srcpad caps are a subset of the template caps	2012-02-01 16:32:53 +01:00
Sebastian Dröge	0370b0dc12	audioencoder: Add gst_audio_encoder_set_output_format() function for consistency	2012-02-01 16:27:47 +01:00
Sebastian Dröge	dbd43c7dd3	audiodecoder: Rename set_outcaps() to set_output_format() and take a GstAudioInfo as parameter	2012-02-01 16:27:47 +01:00
Wim Taymans	30af2fe7d6	audiosrc: wait on the right cond variable ... This broke with a merge commit	2012-01-27 18:27:26 +01:00
Wim Taymans	fcdc385aa1	port to new map API	2012-01-25 12:30:53 +01:00
Sebastian Dröge	68c0790817	Merge branch 'master' into 0.11 ... Conflicts: gst-libs/gst/interfaces/propertyprobe.c sys/xvimage/xvimagesink.c	2012-01-25 11:50:54 +01:00
Wim Taymans	3d42f0f6ed	port to new glib thread API	2012-01-19 11:36:17 +01:00
Tim-Philipp Müller	576bbb4fd8	Remove compatibility code cruft for old GLib versions	2012-01-18 17:22:21 +00:00
Mark Nauwelaerts	3e312e6e16	baseaudiosink: commit correct number of samples when not syncing	2012-01-17 21:46:58 +01:00
Mark Nauwelaerts	974c678ec8	audiodecoder: register state change function	2012-01-17 11:53:51 +01:00
Sebastian Dröge	de19cfdd8a	audio: More UNPOSITION flag sanity checks ... ..and turn the GST_WARNING() into a g_warning(). This is a programming error and should be fixed.	2012-01-11 10:49:49 +01:00
Sebastian Dröge	a03f70e3cd	audio: Add validity check for the UNPOSITIONED audio flag ... Also reset the flag when parsing caps.	2012-01-11 10:44:37 +01:00
Sebastian Dröge	05beab5382	audiometa: Improve GstAudioDownmixMeta to be actually usable ... This now has a two-dimensional array of coefficients as required and also stores the source and destination channel positions.	2012-01-10 12:46:05 +01:00
Sebastian Dröge	67c8b0dfbd	audio: Don't crash if NULL positions are passed to gst_audio_info_set_format()	2012-01-10 12:02:56 +01:00
Sebastian Dröge	5cb3d75dbf	audiobasesink: Fix infinite recursion by chaining up to the correct parent class vfunc	2012-01-09 14:19:54 +01:00
Sebastian Dröge	bb3eb93ee9	audio: Don't check for channel positions in valid order when converting to a channel mask	2012-01-09 08:24:23 +01:00
Edward Hervey	82da418201	audio: Fix size check ... We fail (and return) if the size is *NOT* a multiple of samples.	2012-01-06 15:14:59 +01:00
Wim Taymans	dd43d0697e	audio: expose API to convert channel array to a mask	2012-01-05 13:59:32 +01:00
Sebastian Dröge	9e072ea844	audio: Improve/fix handling of NONE layouts	2012-01-05 10:34:25 +01:00
Sebastian Dröge	8dcea5d498	audio: Add support again for more than 64 channels with NONE layouts	2012-01-05 10:34:25 +01:00
Sebastian Dröge	31c9f7d09a	audio: Fix GST_AUDIO_CHANNEL_POSITION_MASK macro	2012-01-05 10:34:25 +01:00
Sebastian Dröge	9d56bf7712	audioencoder: Proxy the channel mask field instead of the old channel-layout field	2012-01-05 10:34:24 +01:00
Sebastian Dröge	8fe5dc53e0	audiocdsrc: Add the layout field to the caps	2012-01-05 10:34:24 +01:00
Sebastian Dröge	810bfec656	audio: Add "layout" field to the raw audio caps ... This can be used to differentiate between interleaved and non-interleaved audio and whatever comes in the future.	2012-01-05 10:34:24 +01:00
Sebastian Dröge	e2c6b8ec4d	audio: Add function to reorder channel positions from any order to the GStreamer order	2012-01-05 10:34:24 +01:00
Sebastian Dröge	bd40936409	audioringbuffer: Use new function to get a channel reordering map	2012-01-05 10:34:24 +01:00
Sebastian Dröge	9e930a1ade	audio: Add documentation for the new functions	2012-01-05 10:34:24 +01:00
Sebastian Dröge	c9c12372a5	audio: Add public functions to check channel positions validity and to get a reorder map	2012-01-05 10:34:24 +01:00
Sebastian Dröge	225238a913	audioringbuffer: Add support for reordering of channels	2012-01-05 10:34:16 +01:00
Sebastian Dröge	c227f5e77e	audio: Add new channel positions and simplify channel expression in the caps ... The available channel positions are all channels from SMPTE 2036-2-2008 (in that order) and DTS Coherent Acoustics, which are basically all 28 channels that currently can appear. The channels are now expressed in the caps as a channel-mask, which describes which of the channels are present, and an optional channel-reorder-map, which must only be used after negotiation for fixated caps. For negotiation only the channel-mask and the channel count is relevant and all elements are expected to handle all reorder maps. Elements that don't can use the new API to reorder an audio buffer from any order to another order. This simplifies negotiation a lot while still having as few reorderings necassary as possible and still allow all kinds of channel layouts.	2012-01-05 10:27:21 +01:00
Wim Taymans	e9eaf17eae	audioencoder: turn assert into a real error ... Post a real error instead of just asserting. Fixes a unit test.	2012-01-02 15:42:39 +01:00
Tim-Philipp Müller	26e612aeda	playback, mixerutils: gst_registry_get_default() -> gst_registry_get()	2012-01-02 14:32:11 +00:00
Wim Taymans	ed6fd4eb2f	audio: add flag for unpositioned layout ... Check if thr layout is explicitly unpositioned and set a flag in the audio info structure.	2012-01-02 15:01:58 +01:00
Tim-Philipp Müller	c3e6e23b85	audio, rtsp: remove private/protected gtk-doc markup for enums ... This confuses glib-mkenums, and is not really useful anyway. https://bugzilla.gnome.org/show_bug.cgi?id=666618	2012-01-02 00:19:57 +00:00
Tim-Philipp Müller	d877ef13f5	docs: make gtk-doc happier	2011-12-30 19:24:09 +00:00
Tim-Philipp Müller	62e5a67376	audiocdsrc: remove some probing-related vfuncs ... GstPropertyProbe was removed, so these aren't actually used and we probably want something different for the new API.	2011-12-30 16:26:47 +00:00
Tim-Philipp Müller	6a85353a92	audiocdsrc: update for GstIndex removal	2011-12-30 16:18:39 +00:00
Tim-Philipp Müller	31890ef59b	audiocdsrc: make private bits private	2011-12-30 16:12:30 +00:00
Edward Hervey	f562a29284	Merge remote-tracking branch 'origin/master' into 0.11 ... Conflicts: ext/theora/gsttheoraenc.c gst-libs/gst/tag/gstexiftag.c gst/adder/gstadder.c gst/adder/gstadder.h gst/playback/gstdecodebin2.c gst/playback/gstsubtitleoverlay.c tests/check/libs/tag.c	2011-12-30 13:21:35 +01:00
Tim-Philipp Müller	3dfdd6be9d	audioringbuffer: rename GST_BUFTYPE_* to GST_AUDIO_RING_BUFFER_FORMAT_TYPE_* ... Bit unwieldy, but more appropriate. Could also be moved into audio.h as GstAudioFormatType.	2011-12-25 21:38:21 +00:00
Tim-Philipp Müller	80095caa40	audioringbuffer: remove unused GstAudioRingBufferSegState enum and field	2011-12-25 21:23:11 +00:00
Mark Nauwelaerts	e3c78ff661	audioencoder: add a few more debug statements	2011-12-22 16:58:37 +01:00
Mark Nauwelaerts	9bfa65b7d3	audiodecoder: tweak documentation	2011-12-22 16:58:34 +01:00
Wim Taymans	ddc05e0ed1	propertyprobe: remove propertyprobe ... Remove the propertyprobe interface Improve docs	2011-12-21 11:58:53 +01:00
Sebastian Dröge	2760dd2068	audiobasesrc: Use guint8 instead of guchar	2011-12-20 14:36:28 +01:00
Sebastian Dröge	338622fe7e	audioringbuffer: Use guint8 instead of guchar	2011-12-20 14:36:28 +01:00
Mark Nauwelaerts	c41f3cbef0	audiodecoder: set a non-zero default maximum tolerated errors ... Whereas the previous default 0 was backwards compatible in that it lead to erroring out immediately upon any error, elements that are really ported and using the base class error macro can be assumed to intend to improve behaviour rather than maintaining the old one. So, make it easy on those and any future one and tolerate some errors by default, as intended. Fixes #666579.	2011-12-20 12:50:18 +01:00
Wim Taymans	7505b7a55c	add audio metadata ... Add some audio metadata to describe a downmix matrix. Add metadata to media type document.	2011-12-20 12:02:25 +01:00
Vincent Penquerc'h	12be1e6fc5	baseaudiosink: fix late buffer leak	2011-12-13 12:55:45 +00:00
Tim-Philipp Müller	fb6d09055a	Merge remote-tracking branch 'origin/master' into 0.11 ... Conflicts: ext/alsa/gstalsadeviceprobe.c ext/alsa/gstalsamixer.c ext/pango/gsttextoverlay.c ext/pango/gsttextoverlay.h gst-libs/gst/audio/gstaudiobasesink.c gst-libs/gst/audio/gstaudioringbuffer.c gst-libs/gst/audio/gstaudiosrc.c gst-libs/gst/video/Makefile.am gst-libs/gst/video/video.c gst/encoding/gststreamcombiner.c gst/encoding/gststreamsplitter.c gst/playback/gstplaybasebin.c gst/playback/gststreamsynchronizer.c gst/playback/gstsubtitleoverlay.c gst/playback/gsturidecodebin.c sys/xvimage/xvimagesink.c tests/examples/Makefile.am win32/common/libgstvideo.def Video overlay composition disabled for now, needs porting to buffer meta.	2011-12-08 01:19:03 +00:00
Wim Taymans	f096b8a8d8	ringbuffer: remove old _full version	2011-12-06 15:06:12 +01:00
Wim Taymans	9e97260c9f	fix for basesrc changes	2011-12-06 13:59:11 +01:00
Tim-Philipp Müller	5440ae3c18	Suppress deprecation warnings in selected files, for g_static_rec_mutex_* mostly ... GStaticRecMutex is part of our API/ABI, not much we can do here in 0.10 for most of these.	2011-12-04 20:50:25 +00:00
Tim-Philipp Müller	0d98aa25b8	Work around deprecated thread API in glib master ... Add private replacements for deprecated functions such as g_mutex_new(), g_mutex_free(), g_cond_new() etc., mostly to avoid the deprecation warnings. We'll change these over to the new API once we depend on glib >= 2.32. Replace g_thread_create() with g_thread_try_new().	2011-12-04 17:16:30 +00:00
Wim Taymans	1225aa9a78	update for basesink event handler changes	2011-12-02 22:24:43 +01:00
Tim-Philipp Müller	177525f89f	Merge remote-tracking branch 'origin/master' into 0.11 ... Conflicts: gst-libs/gst/netbuffer/gstnetbuffer.c gst/ffmpegcolorspace/avcodec.h gst/ffmpegcolorspace/gstffmpegcodecmap.c gst/ffmpegcolorspace/imgconvert.c gst/ffmpegcolorspace/imgconvert_template.h gst/ffmpegcolorspace/mem.c gst/playback/README gst/playback/gstplaybasebin.c gst/playback/gstplaybasebin.h gst/playback/gstplaybin.c sys/v4l/v4lmjpegsrc_calls.c sys/v4l/videodev_mjpeg.h tests/check/elements/gnomevfssink.c	2011-12-02 11:10:17 +00:00
Piotr Fusik	14644457b0	various: typo fixes ... Fix typos in code and docs. Fixes. #658984	2011-12-02 12:03:27 +01:00
Wim Taymans	59113af604	Use the new GstSample for snapshots ... Make appsink return a GstSample. Remove the pull_buffer_list method because it is not very useful anymore. Pass GstSample to the conversion function. Update playbin2 and examples	2011-12-01 16:53:11 +01:00
Edward Hervey	e44db979f9	audio: Add audio-marshal.list to dist-ed files	2011-11-30 11:33:41 +01:00
Wim Taymans	47cbb230e9	audio: move audio interfaces ... Move the audio related interfaces to the audio library.	2011-11-30 07:57:02 +01:00
Tim-Philipp Müller	0c056a04fe	Merge commit '4a58223e4c824fedc024af435337a769e8ce593e' into 0.11	2011-11-28 21:20:10 +00:00
Wim Taymans	5b868bd424	Update for indexable change	2011-11-28 18:24:03 +01:00
Wim Taymans	468d1dde89	audio: update for clock provider API change	2011-11-28 17:51:41 +01:00
Mark Nauwelaerts	4a58223e4c	audioencoder: elaborate some documentation	2011-11-28 11:37:33 +01:00
Mark Nauwelaerts	9f57d91137	audiodecoder: add some documentation	2011-11-28 11:37:27 +01:00
Mark Nauwelaerts	856a5dd581	audiodecoder: really discard NULL decoded frame altogether ... ... including any timestamp, rather than having that one influence base_ts.	2011-11-28 11:37:23 +01:00
Tim-Philipp Müller	32b14c6ed3	Merge remote-tracking branch 'origin/master' into 0.11 ... Conflicts: ext/vorbis/gstvorbisenc.c gst/playback/gstdecodebin2.c gst/playback/gstplaysinkconvertbin.c gst/videorate/gstvideorate.c	2011-11-26 12:12:59 +00:00
Tim-Philipp Müller	a0639dad38	audio: remove unstable API guards from the audio decoder and encoder base classes	2011-11-25 13:11:54 +00:00
Matej Knopp	817f39608c	Fix printf format compiler warnings for OSX / 64bit ... https://bugzilla.gnome.org/show_bug.cgi?id=662607	2011-11-22 01:00:59 +00:00
Wim Taymans	8fc2a21775	update for activation changes	2011-11-21 13:35:34 +01:00
Wim Taymans	d0bd5f04c0	update for new scheduling query	2011-11-18 17:58:58 +01:00
Wim Taymans	1ad4d20607	add parent to activate functions	2011-11-18 13:56:04 +01:00
Wim Taymans	285702a1a6	fix for scheduling mode rename	2011-11-18 12:37:10 +01:00
Wim Taymans	7afdff3575	Merge branch 'master' into 0.11 ... Conflicts: gst-libs/gst/audio/gstaudiodecoder.c	2011-11-17 17:07:41 +01:00
Wim Taymans	e302833e65	add parent to pad functions	2011-11-17 12:48:25 +01:00
Mark Nauwelaerts	69c2c46472	audioencoder: invalidate format info when setup negotiation failed ... ... which ensures nothing subsequently tries to slip past _chain and into a possibly improperly setup subclass.	2011-11-16 19:03:47 +01:00
Vincent Penquerc'h	f17f918b75	audiodecoder: accept dropped buffers before we know the format ... This allows flacdec to not emit audio for headers, while allowing the base audio decoder to keep its timestamps in sync.	2011-11-16 16:54:03 +00:00
Wim Taymans	2202511e77	add parent to query function	2011-11-16 17:25:17 +01:00
Wim Taymans	28157e6f21	_query_peer_*() -> _peer_query_*()	2011-11-15 18:04:17 +01:00
Wim Taymans	ab9ffa93f5	change getcaps to query ... Add sink and src event functions in rtpbasepayload Add query vmethod to rtpbasepayload.	2011-11-15 18:04:16 +01:00
Vincent Penquerc'h	3e095382a1	audiodecoder: accept dropped buffers before we know the format ... This allows flacdec to not emit audio for headers, while allowing the base audio decoder to keep its timestamps in sync.	2011-11-15 13:29:31 +00:00
Robert Swain	a23dff1fbb	audio: Remove some unused variables	2011-11-14 12:49:50 +01:00
Mark Nauwelaerts	38615abdd8	audiodecoder: improve reverse playback ... ... by doing some more (reverse) timestamp interpolating and refactoring downstream pushing. Fixes #661983.	2011-11-14 12:00:06 +01:00
Tim-Philipp Müller	c76e5804b3	Update for GstURIHandler get_protocols() changes	2011-11-13 23:44:23 +00:00
Tim-Philipp Müller	455f337e3d	gio, appsrc, appsink, cdaudiosrc: update for GstURIHandler API changes	2011-11-13 18:22:06 +00:00
Tim-Philipp Müller	4b0dce5148	Merge remote-tracking branch 'origin/master' into 0.11 ... Conflicts: gst-libs/gst/audio/Makefile.am gst-libs/gst/audio/audio.h tests/examples/seek/jsseek.c tests/examples/seek/seek.c tests/icles/test-colorkey.c	2011-11-13 13:36:29 +00:00
Tim-Philipp Müller	cd21e69913	audio: add GST_AUDIO_INFO_IS_VALID macro and use in audio decoder base class ... API: GST_AUDIO_INFO_IS_VALID	2011-11-13 13:18:16 +00:00
Tim-Philipp Müller	394b1f8c3c	audio: fix order in LIBADD ... Local libs must come first.	2011-11-12 12:13:05 +00:00
Tim-Philipp Müller	756c9e2948	audio: fix order in LIBADD ... Local libs must come first.	2011-11-12 11:58:59 +00:00
Tim-Philipp Müller	dfc13ec632	cdda: rename GstCddaBaseSrc to GstAudioCdSrc and move to libgstaudio ... Another mini-lib down, to make space for new mini libs. Remove bogus copyright line while at it.	2011-11-12 11:58:58 +00:00
Wim Taymans	c42e257751	audio: fix docs	2011-11-11 19:13:52 +01:00
Wim Taymans	b645287775	audio: fix headers ... Add const to some methods. Add padding. Add GType for GstAudioInfo and GstAudioFormatInfo. Add new/copy/free for GstAudioInfo.	2011-11-11 17:53:03 +01:00
Wim Taymans	a3416bc11f	rename baseaudio* -> audiobase*	2011-11-11 12:00:52 +01:00
Wim Taymans	ee7072fe7e	rename GstBaseAudio* ->GstAudioBase*	2011-11-11 11:52:47 +01:00
Wim Taymans	3d0ac3ded2	rename files to match contained objects	2011-11-11 11:33:15 +01:00
Wim Taymans	6511f36fdb	audio: GstRingBuffer -> GstAudioRingBuffer	2011-11-11 11:21:41 +01:00
Wim Taymans	b81af23992	audio: rename internal audio ringbuffer	2011-11-11 10:54:39 +01:00
Wim Taymans	ad8f694ec6	remove bogus files ... They got somehow commited in 7012e88090	2011-11-11 10:39:52 +01:00
Wim Taymans	e338792ab0	update for adapter api changes	2011-11-10 18:32:39 +01:00
Wim Taymans	f8ef57ca48	Merge branch 'master' into 0.11	2011-11-10 17:26:12 +01:00
Vincent Penquerc'h	0d47c615ad	baseaudiosink: make unsigned properties unsigned, not signed	2011-11-10 15:55:31 +00:00
Wim Taymans	57eaf388e0	audio: fix base class vmethods	2011-11-10 16:24:12 +01:00
Wim Taymans	ea9bc40bf9	audiosrc: avoid deadlock	2011-11-10 16:05:19 +01:00
Wim Taymans	1f8fe283f6	audioclock: remove _full version	2011-11-10 13:51:23 +01:00
Wim Taymans	d77c8cafee	Merge branch 'master' into 0.11 ... Conflicts: common ext/pango/gsttextoverlay.c gst-libs/gst/video/video.c	2011-11-09 12:11:59 +01:00
Wim Taymans	372b9329b9	remove query types	2011-11-09 11:47:54 +01:00
Tim-Philipp Müller	d7fc45f42e	docs: fix up some Since: markers	2011-11-07 23:05:44 +00:00
Wim Taymans	7ac25e9b26	Merge branch 'master' into 0.11 ... Conflicts: common configure.ac gst-libs/gst/audio/gstbaseaudiosink.c gst/playback/gstdecodebin2.c gst/playback/gstplaysinkaudioconvert.c gst/playback/gstplaysinkaudioconvert.h gst/playback/gstplaysinkvideoconvert.c gst/playback/gstplaysinkvideoconvert.h	2011-11-07 12:23:15 +01:00
Felipe Contreras	3df415d4c7	baseaudiosink: make discont-wait configurable ... Now we can configure how much time to wait before deciding that a discont has happened. Also, adds getter and setter to allow derived implementations to set this value upon construction. Suggestions and several improvements by Havard Graff. Signed-off-by: Felipe Contreras <felipe.contreras@gmail.com>	2011-11-07 11:58:46 +01:00
Felipe Contreras	0a111bf26e	baseaudiosink: delay the resyncing of timestamp vs ringbuffertime ... A common problem for audio-playback is that the timestamps might not be completely linear. This is specially common when doing streaming over a network, where you can have jittery and/or bursty packettransmission, which again will often be reflected on the buffertimestamps. Now, the current implementation have a threshold that says how far the buffertimestamp is allowed o drift from the ideal aligned time in the ringbuffer. This was an instant reaction, and ment that if one buffer arrived with a timestamp that would breach the drift-tolerance, a resync would take place, and the result would be an audible gap for the listener. The annoying thing would be that in the case of a "timestamp-outlier", you would first resync one way, say +100ms, and then, if the next timestamp was "back on track", you would end up resyncing the other way (-100ms) So in fact, when you had only one buffer with slightly off timestamping, you would end up with *two* audible gaps. This is the problem this patch addresses. The way to "fix" this problem with the previous implementation, would have been to increase the "drift-tolerance" to a value that was greater than the largest timestamp-outlier one would normally expect. The big problem with this approach, however, is that it will allow normal operations with a huge offset timestamp vs running-time, which is detrimental to lip-sync. If the drift-tolerance is set to 200ms, it basically means that lip-sync can easily end up being off by that much. This patch will basically start a timer when the first breach of drift-tolerance is detected. If any following timestamp for the next n nanoseconds gets "back on track" within the threshold, it has basically eliminated the effect of an outlier, and the timer is stopped. If, however, all timestamps within this time-limit are breaching the threshold, we are probably facing a more permanent offset in the timestamps, and a resync is allowed to happen. So basically this patch offers something as rare as both higher accuracy, it terms of allowing smaller drift-tolerances, as well as much smoother, less glitchy playback! Commit message and improvments by Havard Graff. Fixes bug #640859.	2011-11-07 11:33:32 +01:00
Felipe Contreras	3f1395afae	baseaudiosink: rename some variables	2011-11-07 11:18:34 +01:00
Felipe Contreras	fbde258be6	baseaudiosink: use gst_util_uint64_scale_int when appropriate ... It's probably safer this way.	2011-11-07 11:11:08 +01:00
Felipe Contreras	369cf3f14a	baseaudiosink: split drift-tolerance into alignment-threshold ... So that drift-tolerance is used for clock slaving resync, and alignment-threshold is for timestamp drift.	2011-11-07 11:10:05 +01:00
Felipe Contreras	58b9818853	baseaudiosink: trivial comment fixes ... Some found by Havard Graff. Signed-off-by: Felipe Contreras <felipe.contreras@gmail.com>	2011-11-07 10:57:56 +01:00
Wim Taymans	2f8292b495	ringbuffer: store bpf in the right variable	2011-11-04 13:21:24 +01:00
Wim Taymans	a5fa136c0b	update for tag API removal	2011-11-02 12:11:16 +01:00
Wim Taymans	5bdfd6d899	structure: fix for api update	2011-11-02 09:04:27 +01:00
Tim-Philipp Müller	b52c5819fb	Update for pad API changes ... GstProbeType, GstProbeReturn and GstActivateMode -> GstPad*	2011-11-01 00:34:28 +00:00
Tim-Philipp Müller	220ccdf275	audioencoder: save audio info parsed in setcaps in encoder context ... Otherwise we'll just error out when the first buffer gets pushed. This is a porting artefact, in 0.10 the infos were allocated on the heap, now we're doing everything with stack-allocated structs.	2011-10-31 14:22:39 +00:00
Tim-Philipp Müller	5ee51e47a1	ext, gst, gst-libs, tests: update for tag list API changes	2011-10-31 14:22:39 +00:00
René Stadler	7eb0985282	audio: remove old C file generated from template ... Not sure how this one got pulled into a merge. In 0.10, it was moved away to gst-template a long time ago. gstaudiofilterexample.c got generated from gstaudiofiltertemplate.c.	2011-10-31 15:19:54 +01:00
Wim Taymans	95281cc306	Merge branch 'master' into 0.11	2011-10-28 16:24:44 +02:00
Mersad Jelacic	d430eb65c5	audiosink: avoid deadlocking audioringbuffer thread ... ... when it goes into wait for ringbuffer starting just after such having been signalled. Fixes #661738.	2011-10-28 14:07:40 +02:00
Wim Taymans	b70275fa10	audiofilter: use BPF for unit_size	2011-10-28 11:37:31 +02:00
René Stadler	9beff28579	audiofilter: fix get_unit_size	2011-10-28 11:24:00 +02:00
René Stadler	5d2154ff4b	audiofilter: init audio info sooner	2011-10-28 11:24:00 +02:00
René Stadler	372cf41a6d	audio, video: init audio/video format info to UNKNOWN format ... This is to prevent e.g. GST_AUDIO_INFO_FORMAT() from crashing on a NULL pointer dereference when used with an unset info.	2011-10-28 11:24:00 +02:00
Wim Taymans	016d036137	Merge branch 'master' into 0.11 ... Conflicts: configure.ac gst-libs/gst/audio/gstbaseaudiosink.c gst/audioconvert/channelmixtest.c gst/playback/gstplaybasebin.c gst/playback/gstsubtitleoverlay.c tests/examples/Makefile.am tests/examples/audio/Makefile.am	2011-10-27 15:44:58 +02:00
Stefan Sauer	53d7d2e966	interfaces: clean up the use of iface and class/klass	2011-10-21 14:46:48 +02:00
Mark Nauwelaerts	981070eb44	audiodecoder: having gather queue contents implies some draining is in order ... ... which ensures e.g. processing and sending last fragment of reverse playback downstream at EOS.	2011-10-19 16:51:09 +02:00
Tim-Philipp Müller	4e59e63ff7	baseaudiosink: fix unused variable compiler warning if debugging in core is disabled ... https://bugzilla.gnome.org/show_bug.cgi?id=660150	2011-10-19 00:32:13 +01:00
Edward Hervey	12a8fff8ac	audio: Add some default channel positions	2011-10-17 12:00:55 +02:00
Edward Hervey	b4858253dc	audio: Properly handle signedness in gst_audio_format_build_integer()	2011-10-17 12:00:16 +02:00
Edward Hervey	45c4a19472	audio: Indent and doc fixes	2011-10-17 11:45:39 +02:00
Wim Taymans	f1088ed647	update for UNEXPECTED -> EOS flowreturn	2011-10-10 11:39:52 +02:00
Tim-Philipp Müller	ab949eebbd	audiodecoder: update to 0.11 API after merge	2011-10-09 16:15:54 +01:00
Tim-Philipp Müller	303dbaf84b	Merge remote-tracking branch 'origin/master' into 0.11 ... Conflicts: tests/check/pipelines/vorbisdec.c tests/check/pipelines/vorbisenc.c	2011-10-09 16:08:36 +01:00
Alessandro Decina	bc6f00becb	audioencoder: fix compile warning	2011-10-09 16:48:18 +02:00
Mark Nauwelaerts	871b1584c9	audioencoder: only resync to upstream upon discont in perfect ts mode ... ... as documented, where discont is marked here if tolerance has been exceeded.	2011-10-08 20:20:10 +02:00
Mark Nauwelaerts	a7ce550d04	audiodecoder: fix timestamp tolerance handling	2011-10-08 20:20:06 +02:00
Mark Nauwelaerts	d8312994aa	audiodecoder: handle empty input by discarding	2011-10-08 20:20:03 +02:00
Wim Taymans	73b894107a	Merge branch 'master' into 0.11 ... Conflicts: ext/vorbis/gstvorbisdec.c ext/vorbis/gstvorbisenc.c ext/vorbis/gstvorbisenc.h gst/audiotestsrc/gstaudiotestsrc.c	2011-10-08 10:19:06 +02:00
Mark Nauwelaerts	37c629fcc6	audioencoder: make upstream queries MT-safe	2011-10-07 14:52:50 +02:00
Mark Nauwelaerts	77069f01b1	audiodecoder: make upstream queries and events MT-safe	2011-10-07 14:52:48 +02:00
Edward Hervey	b8219faa90	audio: Make sure 'channels' and 'channel-positions' are coherent ... If channel-positions are present, check they match the reported 'channels' value.	2011-10-05 11:57:54 +02:00
Edward Hervey	70d967da7c	audio: Fix overread in channel positions ... The array we're writing to is limited to 64 ... but the amount of input positions might be lower than 64. Therefore use MIN and not MAX to know how many values to read from the array.	2011-10-05 11:51:07 +02:00
Tim-Philipp Müller	6ec5fc8d95	audio: don't use GST_PTR_FORMAT for segments ... Avoids crashes with debugging output enabled.	2011-09-30 10:56:02 +01:00
Wim Taymans	1395378575	audiodecoder: fix refcounting error	2011-09-28 16:08:14 +02:00
Wim Taymans	ca6ebee870	ringbuffer: store info so we can debug it	2011-09-28 16:07:53 +02:00
Wim Taymans	f97a9bdc68	Merge branch 'master' into 0.11	2011-09-28 15:46:40 +02:00
Mark Nauwelaerts	8633eb391d	audiodecoder: really push pending events	2011-09-28 15:42:46 +02:00
Wim Taymans	19626cf27a	audiodecoder: add method to set output caps ... Add a method to configure the output caps. Subclasses can't use gst_pad_set_caps() anymore because then we won't see the caps. Unbreak the padtemplate registration, the GTypeClass that is configured in the object during _init is not the right one, we need to use the klass passed as the argument to the init function..	2011-09-28 15:35:56 +02:00
Tim-Philipp Müller	e4e2e3c7b0	audioencoder: remove more tags from upstream tag events such as bitrate tags ... We want to remove all codec specific tags.	2011-09-28 14:32:20 +01:00
Wim Taymans	19346c2c3b	Merge branch 'master' into 0.11 ... Conflicts: gst-libs/gst/audio/gstaudioencoder.c gst/playback/gstplaybin2.c gst/videotestsrc/videotestsrc.c	2011-09-28 11:35:46 +02:00
Mark Nauwelaerts	01d27ee084	audioencoder: only got_data if we really got some ... ... which avoids going loopy with casual subclass.	2011-09-27 16:58:44 +02:00
Mark Nauwelaerts	24d71cf7a6	audioencoder: really push pending events	2011-09-27 16:58:41 +02:00
Mark Nauwelaerts	803b65613b	audioencoder: send tag event after pending events ... ... which probably includes a pending newsegment event.	2011-09-27 16:21:55 +02:00
Mark Nauwelaerts	89f6720545	audioencoder: protect pending_events with proper lock	2011-09-27 16:21:45 +02:00
Mark Nauwelaerts	9a9541ff35	audioencoder: clean up some documentation	2011-09-27 16:21:41 +02:00
Wim Taymans	4bf9022e0c	docs: improve docs	2011-09-27 11:19:24 +02:00
Wim Taymans	c290b8044a	audioenc: fix compilation	2011-09-26 21:11:14 +02:00
Wim Taymans	f71511edd2	Merge branch 'master' into 0.11 ... Conflicts: gst-libs/gst/audio/gstaudiodecoder.c gst-libs/gst/audio/gstaudioencoder.c gst/encoding/gstencodebin.c	2011-09-26 19:22:05 +02:00
Sebastian Dröge	e4c895dfaf	audioencoder: Improve set_frame_sample_{min,max} documentation	2011-09-26 16:35:55 +02:00
Sebastian Dröge	b767be2f68	audiodecoder: Fix thread safety issues if both pads have different streaming threads	2011-09-26 16:22:00 +02:00
Sebastian Dröge	d0bf465248	audiodecoder: Delay sending of serialized events to finish_frame()	2011-09-26 16:19:42 +02:00
Sebastian Dröge	f3f416004f	Revert "audioencoder: Use GST_BOILERPLATE instead of custom GObject boilerplate code" ... This reverts commit 11e375486e. GST_BOILERPLATE() can't define an abstract type and G_DEFINE_ABSTRACT_TYPE() does not pass the class struct to the instance_init function and there's no way to get the class struct of the current type in instance_init().	2011-09-26 16:02:51 +02:00
Sebastian Dröge	4fa9749106	audioencoder: Add support for requesting a minimum and maximum number of samples per frame ... This extends the special case of a fixed number of samples per frame that was supported before already.	2011-09-26 15:59:22 +02:00
Sebastian Dröge	16c3d6b3d5	audioencoder: Fix thread safety issues if both pads have different streaming threads	2011-09-26 15:45:40 +02:00
Sebastian Dröge	61ffd7cb42	audioencoder: Delay sending of serialized events to finish_frame() ... This makes sure that the caps are already set before any serialized events are sent downstream.	2011-09-26 15:42:14 +02:00
Sebastian Dröge	11e375486e	audioencoder: Use GST_BOILERPLATE instead of custom GObject boilerplate code	2011-09-26 15:34:54 +02:00
Mark Nauwelaerts	abafb030ac	audioencoder: add some tag handling convenience help	2011-09-26 15:15:03 +02:00
Mark Nauwelaerts	a99b313c26	audioencoder: provide CODEC/AUDIO_CODEC handling	2011-09-26 15:10:08 +02:00
Mark Nauwelaerts	aae0312e10	audioencoder: filter AUDIO_CODEC/CODEC tags from passing tag events	2011-09-26 15:10:06 +02:00
Edward Hervey	17bfba09f1	Merge branch 'master' into 0.11 ... Conflicts: ext/ogg/gstoggdemux.c ext/pango/gsttextoverlay.c gst-libs/gst/audio/gstaudioencoder.c gst-libs/gst/audio/gstbaseaudiosrc.c gst/playback/gstsubtitleoverlay.c gst/videorate/gstvideorate.c	2011-09-23 18:27:11 +02:00
Edward Hervey	3f45eb1cfc	gst-libs: Temporarily remove dependency of gstaudio on gstpbutils ... Also re-order the SUBDIRS in the higher-level Makefile so it cleanly installs. https://bugzilla.gnome.org/show_bug.cgi?id=657675	2011-09-23 16:17:45 +02:00
Mark Nauwelaerts	001b4a0072	audioencoder: proxy some more optional downstream caps fields to upstream	2011-09-22 15:47:06 +02:00
Mark Nauwelaerts	2a362a95f7	audioencoder: changed is verily the opposite of equal	2011-09-22 15:47:06 +02:00
Mark Nauwelaerts	b420dd54ea	audioencoder: prevent crashing when comparing to a freshly inited GstAudioInfo	2011-09-22 15:46:56 +02:00
Mark Nauwelaerts	7fa7de9221	audio: some more accessor macros for GstAudioInfo	2011-09-22 15:45:05 +02:00
Mark Nauwelaerts	b44978befe	audiodecoder: fix documentation typo	2011-09-22 15:45:01 +02:00
Tim-Philipp Müller	55182ed841	baseaudiosrc: don't try to fixate "width" field for alaw/mulaw ... Fixes warning when trying to fixate e.g. pulsesrc ! audio/x-alaw ! fakesink.	2011-09-10 18:30:55 +01:00
Tim-Philipp Müller	4529c6dc32	Merge remote-tracking branch 'origin/master' into 0.11 ... Merge in doc updates for audio enums from 0.10, and get rid of the #if #else in the enum list, since that confuses gtk-doc. Conflicts: gst-libs/gst/audio/audio.c gst-libs/gst/audio/audio.h	2011-09-06 16:42:42 +01:00
Wim Taymans	dc28bd1b63	audio: rename IS_LE/BE to _IS_LITTLE_ENDIAN/BIG_ENDIAN	2011-09-06 16:27:27 +01:00
Wim Taymans	f04b8fd8af	audio/video add descriptions ... Add a description to the audio and video format info in case we want to use this later.	2011-09-06 16:46:48 +02:00
Tim-Philipp Müller	36a75bdb71	audio: update internal silent sample defines as well to match 0.11	2011-09-06 15:46:45 +01:00
Wim Taymans	c0d31dd555	rename IS_LE/BE to _IS_LITTLE_ENDIAN/BIG_ENDIAN	2011-09-06 16:46:02 +02:00
Tim-Philipp Müller	91d1112360	audio: update audio format enums to match changes in 0.11 ... And add new audio format info stuff to docs.	2011-09-06 15:36:51 +01:00
Wim Taymans	7012e88090	Merge branch 'master' into 0.11 ... Conflicts: gst-libs/gst/audio/audio.h gst-libs/gst/audio/gstaudiodecoder.c gst-libs/gst/audio/gstaudiodecoder.h gst-libs/gst/audio/gstaudioencoder.c gst-libs/gst/audio/gstbaseaudioencoder.h gst/playback/Makefile.am gst/playback/gstplaybin.c gst/playback/gstplaysink.c gst/playback/gstplaysinkvideoconvert.c gst/playback/gstsubtitleoverlay.c gst/videorate/gstvideorate.c gst/videoscale/gstvideoscale.c win32/common/libgstaudio.def	2011-09-06 15:24:32 +02:00
Wim Taymans	33196cdd2c	audio: change audio format syntax a little ... Remove the _ in front of the endianness prefix. Remove the _3 postfix for the 24 bits formats. Add a _32 postfix after the formats that occupy extra space beyond their natural size. The result is that the GST_AUDIO_NE() macro can simply append the endianness after all formats and that we only specify a different sample width when it is different from the natural size of the sample. This makes things more consistent and follows the pulseaudio conventions instead of the alsa ones.	2011-09-06 12:06:39 +02:00
Tim-Philipp Müller	9a8a989a22	docs: more docs clean-ups	2011-09-06 10:07:33 +01:00
Tim-Philipp Müller	5e61db25b5	audio: fix GST_AUDIO_FORMAT_INFO_IS_*() macros to return a boolean	2011-09-05 23:28:20 +01:00
Tim-Philipp Müller	ba05716485	docs: some docs love	2011-09-05 23:28:20 +01:00
Tim-Philipp Müller	7563e0c9cf	docs: add GstAudioDecoder and GstAudioEncoder to documentation	2011-09-05 23:28:20 +01:00
Tim-Philipp Müller	86e6343759	audio: rename GstBaseAudioDecoder/Encoder to GstAudioDecoder/Encoder ... API: gst_gst_audio_decoder_finish_frame() API: gst_gst_audio_decoder_get_audio_info() API: gst_gst_audio_decoder_get_byte_time() API: gst_gst_audio_decoder_get_delay() API: gst_gst_audio_decoder_get_latency() API: gst_gst_audio_decoder_get_max_errors() API: gst_gst_audio_decoder_get_min_latenc()y API: gst_gst_audio_decoder_get_parse_state() API: gst_gst_audio_decoder_get_plc() API: gst_gst_audio_decoder_get_plc_aware() API: gst_gst_audio_decoder_get_tolerance() API: gst_gst_audio_decoder_get_type() API: gst_gst_audio_decoder_set_byte_time() API: gst_gst_audio_decoder_set_latency() API: gst_gst_audio_decoder_set_max_errors() API: gst_gst_audio_decoder_set_min_latency() API: gst_gst_audio_decoder_set_plc() API: gst_gst_audio_decoder_set_plc_aware() API: gst_gst_audio_decoder_set_tolerance() API: gst_gst_audio_encoder_finish_frame() API: gst_gst_audio_encoder_get_audio_info() API: gst_gst_audio_encoder_get_frame_max() API: gst_gst_audio_encoder_get_frame_samples() API: gst_gst_audio_encoder_get_hard_resync() API: gst_gst_audio_encoder_get_latency() API: gst_gst_audio_encoder_get_lookahead() API: gst_gst_audio_encoder_get_mark_granule() API: gst_gst_audio_encoder_get_perfect_timestamp() API: gst_gst_audio_encoder_get_tolerance() API: gst_gst_audio_encoder_get_type() API: gst_gst_audio_encoder_proxy_getcaps() API: gst_gst_audio_encoder_set_frame_max() API: gst_gst_audio_encoder_set_frame_samples() API: gst_gst_audio_encoder_set_hard_resync() API: gst_gst_audio_encoder_set_latency() API: gst_gst_audio_encoder_set_lookahead() API: gst_gst_audio_encoder_set_mark_granule() API: gst_gst_audio_encoder_set_perfect_timestamp() API: gst_gst_audio_encoder_set_tolerance() https://bugzilla.gnome.org/show_bug.cgi?id=642690	2011-09-05 23:28:13 +01:00
Wim Taymans	e694528155	base: port to 0.11	2011-08-29 13:28:08 +02:00
Wim Taymans	057aecc34e	audio: fix after merge	2011-08-29 11:42:35 +02:00
Wim Taymans	e1287b97ab	Merge branch 'master' into 0.11 ... Conflicts: ext/ogg/gstoggmux.c gst-libs/gst/audio/audio.c gst-libs/gst/audio/audio.h gst-libs/gst/audio/multichannel.h gst-libs/gst/pbutils/Makefile.am gst-libs/gst/pbutils/gstdiscoverer.c gst/playback/gstplaysinkaudioconvert.c gst/playback/gstplaysinkvideoconvert.c win32/common/libgstaudio.def	2011-08-29 11:37:36 +02:00
Tim-Philipp Müller	517153e85a	audio: add GstBaseAudioDecoder and GstBaseAudioEncoder to build ... However, libgstaudio now depends on libgstvideo (via pbutils). https://bugzilla.gnome.org/show_bug.cgi?id=642690 API: gst_audio_info_clear() API: gst_audio_info_convert() API: gst_audio_info_copy() API: gst_audio_info_free() API: gst_audio_info_from_caps() API: gst_audio_info_init() API: gst_audio_info_to_caps() API: gst_base_audio_decoder_finish_frame() API: gst_base_audio_decoder_get_audio_info() API: gst_base_audio_decoder_get_byte_time() API: gst_base_audio_decoder_get_delay() API: gst_base_audio_decoder_get_latency() API: gst_base_audio_decoder_get_max_errors() API: gst_base_audio_decoder_get_min_latency() API: gst_base_audio_decoder_get_parse_state() API: gst_base_audio_decoder_get_plc() API: gst_base_audio_decoder_get_plc_aware() API: gst_base_audio_decoder_get_tolerance() API: gst_base_audio_decoder_get_type() API: gst_base_audio_decoder_set_byte_time() API: gst_base_audio_decoder_set_latency() API: gst_base_audio_decoder_set_max_errors() API: gst_base_audio_decoder_set_min_latency() API: gst_base_audio_decoder_set_plc() API: gst_base_audio_decoder_set_plc_aware() API: gst_base_audio_decoder_set_tolerance() API: gst_base_audio_encoder_finish_frame() API: gst_base_audio_encoder_get_audio_info() API: gst_base_audio_encoder_get_frame_max() API: gst_base_audio_encoder_get_frame_samples() API: gst_base_audio_encoder_get_hard_resync() API: gst_base_audio_encoder_get_latency() API: gst_base_audio_encoder_get_lookahead() API: gst_base_audio_encoder_get_mark_granule() API: gst_base_audio_encoder_get_perfect_timestamp() API: gst_base_audio_encoder_get_tolerance() API: gst_base_audio_encoder_get_type() API: gst_base_audio_encoder_proxy_getcaps() API: gst_base_audio_encoder_set_frame_max() API: gst_base_audio_encoder_set_frame_samples() API: gst_base_audio_encoder_set_hard_resync() API: gst_base_audio_encoder_set_latency() API: gst_base_audio_encoder_set_lookahead() API: gst_base_audio_encoder_set_mark_granule() API: gst_base_audio_encoder_set_perfect_timestamp() API: gst_base_audio_encoder_set_tolerance()	2011-08-27 14:47:50 +01:00
Tim-Philipp Müller	58f515f06a	docs: add since markers to baseaudio{decoder,encoder} documentation	2011-08-27 14:47:50 +01:00
Tim-Philipp Müller	90e3d25891	baseaudiodecoder, baseaudioencoder: fix some compiler warnings ... Leaving the GST_USE_UNSTABLE_API guards in until some of the ported decoders have been updated and it's clear that I didn't mess up anywhere porting things to the new audio API.	2011-08-27 14:47:49 +01:00
Tim-Philipp Müller	52ecb383d7	baseaudioutils: remove, merged into or superseded by audio.c	2011-08-27 14:47:49 +01:00
Tim-Philipp Müller	7f0c7e5f82	baseaudioencoder: port to new GstAudioInfo API	2011-08-27 14:47:49 +01:00
Tim-Philipp Müller	c89b49bfaf	baseaudiodecoder: port to GstAudioInfo API	2011-08-27 14:47:49 +01:00
Tim-Philipp Müller	946ddb6462	audio: add gst_audio_info_{init,clear} and gst_audio_info_{copy,free}	2011-08-27 14:47:49 +01:00
Tim-Philipp Müller	63a3d360dc	audio: add GstAudioFormat, GstAudioFormatInfo and GstAudioInfo ... Same as in 0.11, but with caps parsing/serialising for 0.10 style caps. Add setting default channel positions.	2011-08-27 14:47:01 +01:00
Mark Nauwelaerts	bf4a28f420	baseaudioencoder: remove leftover experimental code	2011-08-27 14:47:00 +01:00
Mark Nauwelaerts	35b172004c	audioutils: modify _parse, add GType support functions	2011-08-27 14:47:00 +01:00
Mark Nauwelaerts	a4d5e33224	baseaudiodecoder: move properties to private storage and add ... _get/_set	2011-08-27 14:47:00 +01:00
Mark Nauwelaerts	7939d37936	baseaudiodecoder: rename property	2011-08-27 14:47:00 +01:00
Mark Nauwelaerts	d71e427c49	baseaudiodecoder: replace context helper structure by various ... _get/_set	2011-08-27 14:47:00 +01:00
Mark Nauwelaerts	a39a66dd4b	baseaudioencoder: move properties to private storage and add ... _get/_set	2011-08-27 14:47:00 +01:00
Mark Nauwelaerts	41a0d6f8f0	baseaudioencoder: rename some properties	2011-08-27 14:47:00 +01:00
Mark Nauwelaerts	6302c9d31d	baseaudioencoder: replace context helper structure by various ... _get/_set	2011-08-27 14:47:00 +01:00
Mark Nauwelaerts	d1ab04f029	baseaudio: rename GstAudioState to GstAudioFormatInfo	2011-08-27 14:47:00 +01:00
Mark Nauwelaerts	ecf57f2b73	baseaudioencoder: TEMP; avoid some imperfect ts jitter ? ... ... even when not in perfect mode ?	2011-08-27 14:47:00 +01:00
Mark Nauwelaerts	5a40343102	baseaudioencoder: debug format fixes	2011-08-27 14:47:00 +01:00
Mark Nauwelaerts	cedbedbbca	baseaudiodecoder: debug format fix	2011-08-27 14:47:00 +01:00
Mark Nauwelaerts	8b6109cdbe	baseaudiodecoder: fixup documentation	2011-08-27 14:47:00 +01:00
Mark Nauwelaerts	5003868dc7	baseaudiodecoder: fix FLUSH_STOP actions	2011-08-27 14:46:59 +01:00
Mark Nauwelaerts	660aa2e2c0	baseaudiodecoder: preserve upstream seek event seqnum	2011-08-27 14:46:59 +01:00
Mark Nauwelaerts	d1f5c34fe7	baseaudioencoder: use buffer running time for granule calculation	2011-08-27 14:46:59 +01:00
Mark Nauwelaerts	6c04035eec	baseaudiodecoder: minor fix in ts resync	2011-08-27 14:46:59 +01:00
Mark Nauwelaerts	d46006b198	baseaudiodecoder: improve glitch resilience ... Provide a replacement for GST_ELEMENT_ERROR to avoid aborting at the first atom out of place, while on the other hand not failing indefinitely.	2011-08-27 14:46:59 +01:00
Mark Nauwelaerts	79b41f59f6	baseaudiodecoder: add limited legacy seeking support	2011-08-27 14:46:59 +01:00
Mark Nauwelaerts	0c33df6540	baseaudiodecoder: cater for audio-codec tag	2011-08-27 14:46:59 +01:00
Mark Nauwelaerts	1dbbe7c89d	baseaudiodecoder: initial version	2011-08-27 14:46:59 +01:00
Mark Nauwelaerts	87409f2587	baseaudioencoder: misc fixes	2011-08-27 14:46:59 +01:00
Mark Nauwelaerts	8c61685554	baseaudio: add audioutils for caps and query handling helper utils	2011-08-27 14:46:59 +01:00
Mark Nauwelaerts	cb04eaaa8f	baseaudioencoder: mark unstable API	2011-08-27 14:46:59 +01:00
Mark Nauwelaerts	b47c08ba17	baseaudioencoder: fix clearing context	2011-08-27 14:46:59 +01:00
Mark Nauwelaerts	e3cae1619c	baseaudioencoder: simplify latency variable handling	2011-08-27 14:46:59 +01:00
Mark Nauwelaerts	9ce2edc918	baseaudioencoder: minor fixes and code simplifications ... Also modify and elaborate a bit on pre_push (though currently unused to no harm).	2011-08-27 14:46:59 +01:00
Mark Nauwelaerts	d0e9fbf3db	baseaudioencoder: additional documentation on granule semantics and ... configuration	2011-08-27 14:46:59 +01:00
Mark Nauwelaerts	9f7849eac9	baseaudioencoder: elaborate property names	2011-08-27 14:46:58 +01:00
Mark Nauwelaerts	bf61f04577	baseaudioencoder: rename state field xint to is_int	2011-08-27 14:46:58 +01:00
Mark Nauwelaerts	3d2f496b3a	baseaudioencoder: gtk-doc syntax fixes	2011-08-27 14:46:58 +01:00
Mark Nauwelaerts	51acb02342	baseaudioencoder: minor fix and cleanup	2011-08-27 14:46:58 +01:00
Mark Nauwelaerts	90d99f23c6	baseaudiocodec: ... and also rename to baseaudiodecoder	2011-08-27 14:46:58 +01:00
Mark Nauwelaerts	dfd7616f60	gst-libs/gst/audio: Remove baseaudiodecoder ... Adds little beyond baseaudiocodec (seeking, bit of query), and what it adds is mainly out-of-scope (e.g. decoder seeking, should be done by upstream demuxer/parser) and/or based on non-prime example (mad).	2011-08-27 14:46:58 +01:00
Iago Toral	492ab47fd2	baseaudiodecoder: Return TRUE if we run into special conversion cases.	2011-08-27 14:46:50 +01:00
Iago Toral	2ed1331f43	audio: initial version of GstBaseAudioCodec ... Moved most of the code to GstBaseAudioCodec, GstBaseAudioDecode is now really small, maybe we do not really need it (or its encoder counterpart). Added more API for subclasses and documentation.	2011-08-27 14:45:47 +01:00
Iago Toral	9740eb35b8	Added src_queries to decoder class. Added handle_discont to decoder ... class. Reworked reset. Various other minor fixes.	2011-08-27 14:45:47 +01:00
Iago Toral	d05c805b16	Added a draft implementation of gstbaseaudiodecoder	2011-08-27 14:45:47 +01:00
Mark Nauwelaerts	fc6b421227	Added audio directory for audio codec base classes	2011-08-27 14:45:47 +01:00
Mark Nauwelaerts	ef92c7438d	audioencoders: add streamheader helper utility	2011-08-27 14:45:47 +01:00
Mark Nauwelaerts	80241fde8d	audioencoders: baseaudioencoder and ported encoders	2011-08-27 14:45:47 +01:00
Wim Taymans	6854f2bbf1	multichannel: add some more channels	2011-08-24 18:39:47 +02:00
Wim Taymans	24ea19935f	audio/video: add format of the pack functions ... Replace the unpack_size with an unpack_format, which is more descriptive of the kind of data the unpack function will create.	2011-08-24 16:40:43 +02:00
Wim Taymans	0a1874461a	audio: rename UNPOSITIONED to DEFAULT_POSITIONS ... Rename the UNPOSITIONED flag to the DEFAULT_POSITIONS flag because that is really what the resulting GstAudioInfo will contain as the chanel mappings.	2011-08-24 14:13:33 +02:00
Wim Taymans	c6758ecfa9	audio: move function to convert	2011-08-22 16:11:27 +02:00
Wim Taymans	3fab57b5cf	Merge branch 'master' into 0.11 ... Conflicts: gst-libs/gst/interfaces/videooverlay.c gst-libs/gst/rtp/gstrtpbuffer.c po/af.po po/az.po po/bg.po po/ca.po po/cs.po po/da.po po/de.po po/el.po po/en_GB.po po/es.po po/eu.po po/fi.po po/fr.po po/gl.po po/hu.po po/id.po po/it.po po/ja.po po/lt.po po/lv.po po/nb.po po/nl.po po/or.po po/pl.po po/pt_BR.po po/ro.po po/ru.po po/sk.po po/sl.po po/sq.po po/sr.po po/sv.po po/tr.po po/uk.po po/vi.po po/zh_CN.po	2011-08-22 13:06:27 +02:00
Stefan Kost	01bbdd6bdf	docs: handle warnings emitted by gtk-doc ... This is useful and in most cases someone had put arbitrary markup into the docs, misspelled xref'ed symbols, forgot to add stuff to the docs etc..	2011-08-20 19:16:42 +02:00
Wim Taymans	0213407fbc	audio: rename INT -> INTEGER ... Spell INTEGER fully instead of using the int abreviation. Remove some old functions.	2011-08-20 10:49:17 +02:00
Wim Taymans	7db6fa37b4	audio: add function to build audio format	2011-08-19 16:00:33 +02:00
Wim Taymans	17dd31b0f4	audio: add more macros	2011-08-19 14:03:23 +02:00
Sebastian Dröge	85a3e7c98c	audiofilter: Pass a const pointer to the audio format info to ::setup() ... It is not meant to be changed by the subclass.	2011-08-19 10:06:39 +02:00
Wim Taymans	dae848818d	audio: rework audio caps. ... Rework the audio caps similar to the video caps. Remove width/depth/endianness/signed fields and replace with a simple string format and media type audio/x-raw. Create a GstAudioInfo and some helper methods to parse caps. Remove duplicate code from the ringbuffer and replace with audio info. Use AudioInfo in the base audio filter class. Port elements to new API.	2011-08-18 19:15:03 +02:00
Wim Taymans	d1a83d7a41	baseaudiosrc: chain up to parent in fixate	2011-08-17 17:24:35 +02:00
Wim Taymans	33467d9629	Merge branch 'master' into 0.11 ... Conflicts: configure.ac ext/pango/gsttextoverlay.c ext/theora/gsttheoradec.c gst/adder/gstadder.c gst/adder/gstadder.h gst/audioresample/gstaudioresample.c gst/encoding/gstencodebin.c gst/playback/gstdecodebin.c gst/playback/gstdecodebin2.c tests/check/elements/decodebin2.c tests/check/elements/playbin-compressed.c win32/common/libgsttag.def	2011-08-16 18:01:14 +02:00
Wim Taymans	d6740006d4	audio: remove deprecated methods	2011-08-16 16:59:15 +02:00
Josep Torra	5629ed74b3	Fix debug statements ... Fixes build on MacOSX Signed-off-by: Edward Hervey <edward.hervey@collabora.co.uk>	2011-08-10 11:15:41 +02:00
Wim Taymans	86a10fbb9f	baseaudiosrc: call parent alloc function ... Call the parent alloc function to allocate buffers.	2011-08-04 18:08:49 +02:00
Stefan Sauer	264d91a502	baseaudiosink: fix latency calculation for live elements ... Max_latency was computed on already adjusted min_latency. Introduce a new variable for clarity. Spotted by Blaise Gassend. Fixes #644284	2011-07-28 14:31:47 +02:00
Mark Nauwelaerts	68231a645a	baseaudiosink: fix max latency calculation ... ... to allow infinite max, as also claimed by comment.	2011-07-28 12:05:06 +02:00
Mark Nauwelaerts	5d0f279fea	baseaudiosink: drop samples that are too late ... ... rather than having all of them rendered at 0 or subsequently aligned, likely inevitably leading to repeated resyncing.	2011-07-28 11:47:52 +02:00
Wim Taymans	a3971d2afe	baseaudiosink: chain up to parent_class correctly	2011-07-26 12:42:22 +02:00
Wim Taymans	8aea5d34bd	baseaudiosink: use new basesink query vmethod	2011-07-26 12:37:04 +02:00
Tim-Philipp Müller	4bf26ba5d2	Add -DGST_USE_UNSTABLE_API to the compiler flags to avoid warnings	2011-07-05 10:07:08 +01:00
Wim Taymans	a58805216a	audio: clean up headers	2011-06-21 18:17:59 +02:00
Wim Taymans	2e837743c3	audio: clean up audiosink headers	2011-06-21 18:13:48 +02:00
Wim Taymans	d9e1e23094	audio: clean up ringbuffer header	2011-06-21 18:08:12 +02:00
Wim Taymans	f9967e4aac	Merge branch 'master' into 0.11 ... Conflicts: gst-libs/gst/video/video.h gst/playback/gstplaysinkaudioconvert.c gst/playback/gstplaysinkvideoconvert.c tests/check/libs/rtp.c	2011-06-02 12:18:13 +02:00
Stefan Kost	940291dd38	audio: move testchannels example to 'tests/examples' dir ... Also fix it up a little to not include 'c' file but link to the libs instead.	2011-05-27 15:09:25 +03:00
Wim Taymans	e614c6bd81	feature: use object name instaed of feature name	2011-05-24 18:21:06 +02:00
Wim Taymans	010add200a	scheduling: port to new scheduling query	2011-05-24 17:37:45 +02:00
Wim Taymans	a87c021237	Merge branch 'master' into 0.11 ... Conflicts: gst-libs/gst/video/convertframe.c	2011-05-24 09:47:15 +02:00
Stefan Kost	6bee2cb4ee	docs: add missing documentation for various pieces	2011-05-23 23:56:09 +03:00
Thijs Vermeir	dad50ad1fe	baseaudiosink: recalibrate clock on setcaps ... Because the spec for the ringbuffer can change when changing the caps, we must recalibrate the clock. https://bugzilla.gnome.org/show_bug.cgi?id=610443	2011-05-23 17:02:03 +02:00
Stefan Kost	089fdb7792	docs: fixup audio-library docs	2011-05-23 15:08:24 +03:00
Stefan Kost	d6ea8d5cb3	docs: fix docs for new api ... Some parameters where wrong, first line missed the ':' and return docs where broken.	2011-05-23 14:56:17 +03:00
Sebastian Dröge	8a0bdbf2bc	audiofilter: gst_pad_template_new() does not take ownership of the caps anymore ... There's no need to copy the caps before passing them to that function.	2011-05-17 12:31:18 +02:00
Sebastian Dröge	318ed07598	Revert "-base_port to new query API" ... This reverts commit c9f4e0676b.	2011-05-17 11:25:31 +02:00
Sebastian Dröge	d0362c2b87	Merge branch 'master' into 0.11 ... Conflicts: configure.ac ext/alsa/gstalsasrc.c gst-libs/gst/audio/gstbaseaudiosink.c gst-libs/gst/tag/gstxmptag.c gst/playback/gstsubtitleoverlay.c gst/videorate/gstvideorate.c sys/xvimage/xvimagesink.c	2011-05-16 17:06:22 +02:00
Wim Taymans	94dfe80f71	-base: port to new SEGMENT API	2011-05-16 13:48:11 +02:00
Arun Raghavan	623e8781ab	baseaudiosink: Use g_str_equal() instead of strncmp() ... The strncmp is unnecessary anyway since one of the strings is a const string.	2011-05-14 18:53:12 +05:30
Arun Raghavan	824e643ec9	baseaudiosink: Fix trivial indentation problems	2011-05-14 18:53:12 +05:30
Arun Raghavan	8ff93a6a3d	audio: Add an IEC 61937 payloading library ... This can be used by sinks to take compressed formats, correctly payload these in IEC 61937 frames and feed these to sinks that support passthrough output over IEC 60958 (S/PDIF) or, in the case of MP3, over Bluetooth. Initial implementation includes AC3, E-AC3, MPEG-1, MPEG-2 (non-AAC), and DTS (type-I/II/II) payloading. More formats can be added as needed. API: gst_audio_iec61937_frame_size() API: gst_audio_iec61937_payload() https://bugzilla.gnome.org/show_bug.cgi?id=642730	2011-05-14 18:53:12 +05:30
Arun Raghavan	643e5f586c	baseaudiosink: Allow subclasses to provide payloaders ... This allows subclasses to provide a "payload" function to prepare buffers for consumption. The immediate use for this is for sinks that can handle compressed formats - parsers are directly connected to the sink, and for formats such as AC3, DTS, and MPEG, IEC 61937 patyloading might be used. API: GstBaseAudioSinkClass:payload() https://bugzilla.gnome.org/show_bug.cgi?id=642730	2011-05-14 18:23:18 +05:30
Arun Raghavan	9615081f9c	ringbuffer: Add support for E-AC3 ... Adds support for pushing E-AC3 buffers and doing bytes-to-ms conversion correctly. The assumption (as with other formats) is that something like IEC 61937 payloading will be used. Correspondingly the ringbuffer spec is populated so that the data rate is 4x normal AC3. https://bugzilla.gnome.org/show_bug.cgi?id=642730	2011-05-14 18:21:23 +05:30
Arun Raghavan	193fbf93a9	ringbuffer: Add support for MPEG audio buffers	2011-05-14 18:21:16 +05:30
Arun Raghavan	1a1f2cc50a	ringbuffer: Add AAC format types ... These are meant to be used for buffers containing AAC data. Nothing uses this yet, but for now it serves to distinguish from GST_BUFTYPE_MPEG which represents non-AAC MPEG audio. API: GST_BUFTYPE_MPEG2_AAC API: GST_BUFTYPE_MPEG4_AAC	2011-05-14 18:20:37 +05:30
Arun Raghavan	33ef9ab054	ringbuffer: Add support for DTS buffers	2011-05-14 16:53:33 +05:30
Wim Taymans	c9f4e0676b	-base_port to new query API	2011-05-10 18:39:07 +02:00
Wim Taymans	816f4e791d	segment: fix for new core API ... Fix for gst_*_segment_full rename.	2011-05-09 18:16:46 +02:00
Wim Taymans	ec57868488	-base: don't use buffer caps ... Port to newest 0.11 core API, remove GST_PAD_CAPS and GST_BUFFER_CAPS.	2011-05-09 13:05:12 +02:00
Sebastian Dröge	68a3828adb	audiofilter: GstElement takes ownership of pad templates and it should be called from class_init now, not base_init	2011-04-19 14:31:20 +02:00
Sebastian Dröge	f50b3af5d7	audio: Use G_DEFINE_TYPE instead of GST_BOILERPLATE	2011-04-19 10:52:00 +02:00
Sebastian Dröge	0759ce8533	Merge branch 'master' into 0.11	2011-04-18 13:23:32 +02:00