Commit graph

1486 commits

Author SHA1 Message Date
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
Håvard Graff
d9f1b3736e ringbuffer: make sure to not start if the may_start flag is FALSE
Fixes #635784
2011-04-18 11:40:06 +02:00
Sebastian Dröge
c8792778f8 Merge branch 'master' into 0.11 2011-04-16 16:06:26 +02:00
Tim-Philipp Müller
1d05e81435 libs: gobject-introspection scanner doesn't need to scan or update plugin info
Make sure the scanner doesn't load or introspect or check any plugins,
(especially not outside the build directory).
2011-04-16 11:01:53 +01:00
Wim Taymans
6e160bed3d Merge branch 'master' into 0.11
Conflicts:
	android/alsa.mk
	android/app.mk
	android/app_plugin.mk
	android/audio.mk
	android/audioconvert.mk
	android/decodebin.mk
	android/decodebin2.mk
	android/gdp.mk
	android/interfaces.mk
	android/netbuffer.mk
	android/pbutils.mk
	android/playbin.mk
	android/queue2.mk
	android/riff.mk
	android/rtp.mk
	android/rtsp.mk
	android/sdp.mk
	android/tag.mk
	android/tcp.mk
	android/typefindfunctions.mk
	android/video.mk
2011-04-11 11:37:51 +02:00
Alessandro Decina
030f639a8e android: make it ready for androgenizer
Remove the android/ top dir
Fixe the Makefile.am to be androgenized

To build gstreamer for android we are now using androgenizer which generates the
needed Android.mk files.
Androgenizer can be found here:
http://git.collabora.co.uk/?p=user/derek/androgenizer.git
2011-04-11 07:23:21 +02:00
Wim Taymans
da1c863711 Merge branch 'master' into 0.11
Conflicts:
	gst-libs/gst/tag/gstvorbistag.c
2011-04-04 11:31:33 +02:00
Stian Johansen
0f8edca902 baseaudiosrc: Add src object lock around call to ringbuffer parse caps.
A race was observed between query() and setcaps() where the latter would
change the ringbuffer spec while the former was performing operations
based this data.
2011-04-04 09:35:58 +02:00
Havard Graff
63cfa2a50d baseaudiosrc: protect against ringbuffer disappearing while in a query
Observed a case where the src went to null-state during the query,
hence the spec pointer was no longer valid, and
gst_util_unit64_scale_int crashed (assertion `denom > 0´failed)

Add locking to make sure the ringbuffer can't disappear.
2011-04-04 09:33:33 +02:00
Havard Graff
588ac0ae6f baseaudiosink: don't allow aligning behind the read-segment
Given a large enough drift-tolerance, one could end up in a situation
where one would keep aligning the written buffers behind the current
read-segment position. The result for the reader would be complete
silence, possible preceded by very choppy audio.

By checking the available headroom, one can determine if there is
room to do alignment, or if one should resort to a resync instead to get
the pointers back on track.

Also refactor the alignment-logic out of the render function for cleaner
code.
2011-04-04 09:31:26 +02:00
Wim Taymans
d96a8c1aa7 Merge branch 'master' into 0.11 2011-03-31 17:53:12 +02:00
Mark Nauwelaerts
e73f293ee5 baseaudiosink: arrange for running clock when rendering eos
Commit ba2e500bd9 ensured to provide
a running clock when EOS had finished rendering.  However,
other measures are needed (and were in place before) to ensure a
running clock when EOS still needs rendering (i.e. waiting).

So, specifically, re-introduce eos_rendering removed in aforementioned commit,
this time as a public variable so subclasses can be aware of the situation.

Fixes (part of) #645961.

API: GstBaseAudioSink:eos_rendering
2011-03-31 13:18:53 +02:00
Tim-Philipp Müller
45b6bda76c libs: make sure gobject-introspection scanner calls gst_init()
Cherry-picked from 0.11, since it's the right thing to do (we
now silently rely on various _get_type() working without
gst_init() having been called).
2011-03-30 21:08:29 +01:00
Tim-Philipp Müller
a818fe7381 libs: replace 0.10 with @GST_MAJORMINOR@ in Makefile.am
For easier cherry-picking/merging later.
2011-03-30 20:57:32 +01:00
Wim Taymans
248ab2d064 Fix for latest API changes 2011-03-30 16:50:45 +02:00
Wim Taymans
536e86e28f tests: fix more checks 2011-03-28 19:23:38 +02:00
Wim Taymans
e6dc4c189d tests: fix some unit tests 2011-03-28 16:54:30 +02:00
Wim Taymans
d10602fbde audiosink: improve comment 2011-03-28 10:25:38 +02:00
Wim Taymans
3d25a4b470 libs: port to new data API 2011-03-27 13:55:15 +02:00
Tim-Philipp Müller
842911d241 libs: make sure gobject-introspection scanner calls gst_init()
Fixes introspection failures caused by type assertions/warnings.
Since we now moved from _get_type() functions to external GType
variables in a couple of places, we actually have to call gst_init()
to make sure these are set when we use GST_TYPE_FOO.
2011-03-09 12:17:14 +00:00
Wim Taymans
8a786d10be baseaudiosink: use sink preroll lock 2011-03-04 17:25:46 +01:00
Wim Taymans
6aa22111a1 Merge branch 'master' into 0.11 2011-03-04 16:21:13 +01:00
Mark Nauwelaerts
ba2e500bd9 baseaudiosink: start ringbuffer upon going to PLAYING and already EOS
... otherwise we may end up without running clock in PLAYING.

Fixes #636886.
2011-03-04 14:10:30 +01:00
Wim Taymans
65ba216b8c baseaudiosink: remove deprecated method 2011-02-28 11:50:03 +01:00
Wim Taymans
c6dd11981d Merge branch 'master' into 0.11
Conflicts:
	configure.ac
	gst-libs/gst/pbutils/Makefile.am
2011-02-28 11:47:44 +01:00
Felipe Contreras
21d1e2ded0 baseaudiosink: trivial cleanups
It seems these stuff was neglected from commmit d8942e2.

Signed-off-by: Felipe Contreras <felipe.contreras@nokia.com>
2011-01-30 15:40:53 +02:00
Tim-Philipp Müller
0ed757db33 gobject-introspection: use same PKG_CONFIG_PATH for g-ir-compiler as for g-ir-scanner
Make sure to use the PKG_CONFIG_PATH set at configure time instead of
just relying on an env-var set one. This makes sure both g-ir-compiler
and g-ir-scanner use the same PKG_CONFIG_PATH for determining include
paths etc.
2011-01-08 02:10:03 +00:00
Tim-Philipp Müller
9c9afee1cf baseaudiosink: default to enable-last-buffer=FALSE for audio sinks
There isn't really any good reason to get the last buffer from an
audio sink, so don't make the sink keep it around unnecessarily.
2011-01-02 17:21:54 +00:00
Havard Graff
60ff7c0eb4 baseaudiosink: protect against ringbuffer disappearing while in a query
Observed a case where the sink went to null-state during the query,
hence the ringbuffer-pointer was NULL, causing a crash.

Moving the ringbuffer-check code until after the query, and hold the
lock during the check and while using the spec-values. It should not matter
to the query wether the ringbuffer is present or not, and it actually
gets a time bit more time to get the ringbuffer set up in this case!

Fixes #635231
2010-12-29 12:29:40 +01:00
Wim Taymans
eee6bc7dc9 more 0.10 -> 0.11 changes 2010-12-06 17:09:10 +01:00
Evan Nemerson
8fb2c27ed0 introspection: Add information on exported packages to GIRs
https://bugzilla.gnome.org/show_bug.cgi?id=635392
2010-11-21 00:44:37 +00:00
Stefan Kost
83c14483ed various: add a missing G_PARAM_STATIC_STRINGS flag to object properties 2010-10-13 16:13:31 +03:00
Tim-Philipp Müller
751c34bffc audio: make public get_type() functions thread-safe 2010-10-08 11:34:58 +01:00
Tim-Philipp Müller
6b7af81e30 audio: fix enum value name in enums that are public API
So run-time bindings can introspect the names correctly (we abuse this
field as description field only in elements, not for public API
(where the description belongs into the gtk-doc chunk).

https://bugzilla.gnome.org/show_bug.cgi?id=629746
2010-10-08 11:34:58 +01:00
Wim Taymans
84dba3698d baseaudiosink: add Since markers
Fixes #630443
2010-09-24 13:09:28 +02:00
Havard Graff
3067a83df2 baseaudiosink: Added getter and setter for drift tolerance. 2010-09-24 13:06:35 +02:00
Wim Taymans
c89082b2dd baseaudiosink: subtract the render_delay from our latency
The latency reported by the base class includes the render_delay, which we don't
want to include when we start slaving our clocks.

See #630441
2010-09-24 12:54:47 +02:00
Sebastian Dröge
550d59354f ringbuffer: Use G_DEFINE_ABSTRACT_TYPE instead of manual GObject boilerplate code
This also makes the _get_type() function threadsafe.

Fixes bug #630440.
2010-09-23 23:58:50 +02:00
Wim Taymans
24226284b8 baseaudio: avoid taking extra ref on sink/src
Don't take an extra ref on the sink and source because that creates a reference
cycle. Instead, use the invalidate method of the clock when the sink and source
are freed. This way, we don't call into the time function anymore after the
objects are disposed.
2010-09-07 18:12:38 +02:00
Wim Taymans
c7972692d3 audioclock: add a function to invalidate the clock
Add a function to invalidate the time function of a clock. Useful for when the
function becomes invalid.
2010-09-07 18:12:38 +02:00
Tim-Philipp Müller
e776699036 build: use new AG_GST_PKG_CONFIG_PATH m4 macro from common
Sets up a GST_PKG_CONFIG_PATH variable for use in Makefile.am
(avoids trailing ':' in PKG_CONFIG_PATH used).
2010-08-14 19:12:37 +01:00
Tim-Philipp Müller
b61b83376a introspection: set PKG_CONFIG_PATH so that our in-tree libs come first when calling scanner
When calling gobject-introspection scanner, make sure our own
freshly-built libs within the source tree (well, build dir) come
first in the PKG_CONFIG_PATH. May or may not help to make sure
that it doesn't pick up older external plugins-base libs (or
.gir files) from outside the source tree / build directory as
dependencies of the introspected lib instead of using the
stuff we just built in a sibling directory.

https://bugzilla.gnome.org/show_bug.cgi?id=623698
2010-08-14 19:11:48 +01:00
Sebastian Dröge
b296c96169 baseaudiosink/baseaudiosrc: Post CLOCK-LOST/CLOCK-PROVIDE when going to/from READY
Otherwise the clocks are redistributed every time the pipeline
goes to PAUSED, which is quite expensive.
2010-08-04 15:19:42 +02:00
Wim Taymans
f9404c0b27 ringbuffer: improve debugging 2010-08-04 10:33:32 +02:00
Wim Taymans
2304ff9095 ringbuffer: whitespace fixes 2010-08-04 10:33:32 +02:00
Sebastian Dröge
ed271ff809 baseaudiosink: Post clock-provide and clock-lost messages when going from/to PLAYING 2010-07-16 17:40:45 +02:00
Sebastian Dröge
e84c7f02b4 baseaudiosrc: Post clock-provide and clock-lost messages when going from/to PLAYING 2010-07-16 17:40:45 +02:00
Sebastian Dröge
f1ac770f1b baseaudiosink: Use new gst_audio_clock_new_full() 2010-07-16 17:40:45 +02:00
Sebastian Dröge
32b0b0aef9 baseaudiosrc: Use new gst_audio_clock_new_full() 2010-07-16 17:40:45 +02:00
Sebastian Dröge
8989ad93d9 audioclock: API: Add gst_audio_clock_new_full() with a GDestroyNotify for the user_data
Elements usually use their own instance as instance data but the
clock can have a longer lifetime than their elements and the clock
doesn't own a reference of the element.

Fixes bug #623807.
2010-07-16 17:40:17 +02:00
Wim Taymans
2ced0a3d5d ringbuffer: check for ringbuffer state first
Check for the state of the ringbuffer before doing the checks of the other
buffer properties, when we're not started, we don't care about those values.
2010-06-25 17:21:57 +02:00
Sebastian Dröge
a5c35621c3 Revert "baseaudiosink: Allocate and free the clock in NULL->READY and reverse"
This reverts commit cea2644ed8.

Many audio sink assume that they can create a clock in
the instance init function and it will be there forever
and not be cleared by the state change functions.
2010-06-03 13:44:40 +02:00
Sebastian Dröge
cea2644ed8 baseaudiosink: Allocate and free the clock in NULL->READY and reverse 2010-06-03 10:23:22 +02:00
Vincent Untz
764c899215 libs: point gobject-introspection scanner to .la files
Point g-ir-scanner to the .la file of our library, which hopefully
makes it find the right dependencies in all cases (ie. our locally
built libgstreamer and not the system-installed one). This is also
how it's done in Gtk+ and how it's documented in the wiki, see
http://live.gnome.org/GObjectIntrospection/AutotoolsIntegration

Fixes #603710.
2010-04-03 14:03:45 +01:00
Tim-Philipp Müller
b37c993e4e gst-libs: more gobject-introspection fixes
Use right .pc file variable for compiler includes this time:
g-ir-compiler wants the girdirs not the typelibdirs as includes.
2010-03-30 23:46:10 +01:00
Tim-Philipp Müller
64cfa6bf73 gst-libs: fix up gobject-introspection some more
Use new girdir and typlibdir from core .pc files, so we can figure
out the right includes to pass to the gobject-introspection tools,
whether core is installed in the same prefix as gobject-introspection
or in a different prefix or uninstalled. This also keeps us from adding
bogus paths to the includes that only work if core is uninstalled.

Also add some missing includes/pkgs where needed.
2010-03-30 19:56:56 +01:00
Tim-Philipp Müller
58a92964c6 build: Makefile.am fixes
Mostly just add missing $(GST_BASE_CFLAGS), but also fix up order
of flags (see docs/random/moving-plugins).
2010-03-19 01:00:36 +00:00
Mark Nauwelaerts
dcc4b25686 baseaudiosink: arrange for a running ringbuffer/clock for _wait_eos
Fixes #612223.
2010-03-16 15:30:12 +01:00
Tim-Philipp Müller
e836151009 docs: more helper libraries docs fixes
Quieten gtk-doc a bit more.
2010-03-16 00:44:50 +00:00
Benjamin Otte
43b1683421 Add -Wmissing-declarations -Wmissing-prototypes to warning flags
Includes all the fixes necessary to make stuff compile again.
2010-03-11 13:50:31 +01:00
Sebastian Dröge
d5a4ca9962 build: Make some more rules silent if requested 2010-03-09 21:01:38 +00:00
Tim-Philipp Müller
e6d868c31c audiosrc: add gratuitious FIXME for use of generic G_TYPE_POINTER type 2010-01-27 00:42:37 +00:00
Sebastian Dröge
6dfc0270ec audio: Use rounding scaling functions for GST_CLOCK_TIME_TO_FRAMES and _FRAMES_TO_CLOCK_TIME
Fixes bug #607381.
2010-01-19 09:26:37 +01:00
Tim-Philipp Müller
848a7f2868 baseaudiosink: increase default drift tolerance to fix glitches with WMA
Increase default drift tolerance to 40ms to avoid glitches with decoders
or formats where there's a lot of timestamp jitter for some reason or
another (in this case: asf/wma), at least until we implement timestamp
smoothing.
2009-12-20 23:19:41 +00:00
Sebastian Dröge
51e2cafe0e audiofilter: Use G_DEFINE_ABSTRACT_TYPE_WITH_CODE
...and fix code style a bit.
2009-11-26 10:38:29 +01:00
Sebastian Dröge
3949cba47d audiofilter: Add _CAST variants of the cast macros 2009-11-26 10:38:28 +01:00
Wim Taymans
75c5aed1ba audiosink: add adjustement when slaving
Our calibration against the pipeline clock is done with the adjusted
ringbuffer time, so take the adjustement into account. Fixes some audio dropouts
when reusing audio sinks after switching clocks and slaving methods in a
pipeline.
2009-11-25 10:26:16 -06:00
Stefan Kost
9e8db533a1 debug: fix format string that was missing a var 2009-11-21 17:47:26 +02:00
Wim Taymans
0e6b9e596d baseaudiosink: fix initial calibration
When we are calibrating the internal clock against the external clock take into
account the time offset applied to our internal clock because we will subtract
that in the render_function again.
2009-11-18 17:11:03 +01:00
Mark Nauwelaerts
0fb680f680 baseaudiosrc: fix 'uninitialized' compiler warning 2009-11-18 12:37:44 +01:00
Wim Taymans
4f3f9a1054 basesrc: fix startup position in the ringbuffer
When we start and we need to produce the first sample, go to the next sample
that will be written into the ringbuffer instead of trying to go to sample 0.
We relied on rather small ringbuffer sizes to correctly go to the current
sample, which breaks whith large buffers.

Fixes #600945
2009-11-06 12:22:00 +01:00
Wim Taymans
d8942e2850 baseaudiosink: make drift tolerance configurable
Add drift-tolerance property (defaulting to 20ms) to handle resync after clock
drift or timestamp drift instead of relying on the latency-time value for clock
drift and 500ms for timestamp drift.
Remove warning about discont timestamp and simply resync. The warning is in some
cases not correct and is triggered more frequently now that we lower the
tolerance value.
2009-11-04 16:16:31 +01:00
Tim-Philipp Müller
6f4c1ac583 Remove GST_DEBUG_FUNCPTR where they're pointless
There's not much point in using GST_DEBUG_FUNCPTR with GObject
virtual functions such as get_property, set_propery, finalize and
dispose, since they'll never be used by anyone anyway. Saves a
few bytes and possibly a sixteenth of a polar bear.
2009-10-28 00:59:35 +00:00
Stefan Kost
f1c32d0fbb build: fix previous commit to fully accomodate the glib-gen.mak changes
I also renamed glib_enum_prefix to glib_gen_prefix as we also use that for the
marshallers. Also rename the rtsp-marshal.list to work with the unified prefix.
2009-10-16 10:56:56 +03:00
Stefan Kost
a89c1de0ea build: use gst-glib-gen.mak to fix the glib build rules. Fixes #598114
The build rules in glib-gen.mak were using pattern rules in a non save way.
2009-10-16 10:23:09 +03:00
Tommi Myöhänen
02cbde648c baseaudiosrc: fix timestamp comparission, Fixes #597407 2009-10-13 19:17:49 +03:00
Wim Taymans
5dbaccabca audioclock: whitespace fixes 2009-10-12 15:47:28 +02:00
Josep Torra
ccec231d2b audio: fix warnings building on macosx 2009-10-09 14:09:02 +02:00
Sebastian Dröge
df9b8b57b3 introspection: Build pkgconfig before all libraries and set PKG_CONFIG_PATH
This way g-ir-scanner can find the gstreamer-*-0.10 pkg-config files.
2009-09-13 11:19:50 +02:00
Tim-Philipp Müller
e4e8417eeb ringbuffer: fix build against core that has debugging disabled
The macro is called GST_DISABLE_GST_DEBUG, not GST_DISABLE_DEBUG.
2009-09-11 10:03:56 +01:00
Stefan Kost
312d7d8014 ringbuffer: add human readable format names when logging
Add string array with human readable names for format and type to be used in log
statements.
2009-09-10 23:01:36 +03:00
Wim Taymans
35cddfb1e3 baseaudiosink: add ugly backward compat hack
Check for pulsesink < 0.10.17 because it includes code that is now included in
baseaudiosink. Disable that code in baseaudiosink to be compatible with the
older version.
2009-09-10 12:40:01 +02:00
Wim Taymans
06be2b8632 baseaudiosink: take clock time in setcaps
Take the time of the clock so that the last_time field is set. This is important
for sinks that restart their internal ringbuffer after a caps change and need to
know the last know position.
2009-09-09 18:26:03 +02:00
Wim Taymans
451789735c audioclock: add some more debug 2009-09-09 18:26:03 +02:00
Wim Taymans
fe47c6c4d5 baseaudiosink: correct for clock reset
When going to NULL, we reset the ringbuffer so that it starts beck from 0. We
also make sure that the clock is updated with the elapsed time so that it
alsways increments even when the ringbuffer goes back to 0. When this happened
we need to adjust the sample position for the reset ringbuffer.

Fixes #594136
2009-09-09 16:19:32 +02:00
Wim Taymans
47550f6984 baseaudiosink: whitespace fixes 2009-09-09 16:17:02 +02:00
Wim Taymans
70f01fd797 ringbuffer: add more debug 2009-09-09 16:16:40 +02:00
Håvard Graff
058776bcf1 baseaudiosrc: improve slave skew resync
The old one did the mistake of not actually advancing the ringbuffer, it just
adjusted the segbase, introducing the whole lenght of the ringbuffer as an
extra delay in the pipeline.

Also make sure that the resync can never go back in time, producing the same
timestamps that has already been produced, as this can cause severe problems
for sinks and other synching mechanisms.

Fixes #594256
2009-09-08 12:59:20 +02:00
Sebastian Dröge
40aba9e0dc introduction: Fix out-of-tree build 2009-09-05 13:46:58 +02:00
Sebastian Dröge
c53499c62b audio: Remove debug echo 2009-09-05 13:09:17 +02:00
Sebastian Dröge
93e19acfec audio: Fix build of introspection data by using dependency order for the headers/sources 2009-09-05 13:08:19 +02:00
Sebastian Dröge
7e90e0846c introspection: Strip Gst prefix from all types/functions 2009-09-05 12:31:47 +02:00
Sebastian Dröge
7794caf9f8 introspection: Fix build if gir-repository is not installed 2009-09-05 11:49:41 +02:00
Sebastian Dröge
d91f5000e1 libs: Add nodist headers and sources to the introspection files 2009-09-05 11:31:48 +02:00
Sebastian Dröge
403f353bba audio: Add gobject-introspection support 2009-09-05 11:09:33 +02:00
Eero Nurkkala
8ad8591e41 ringbuffer: Improve audiosink startup performance
When we start the ringbuffer, immediatly continue processing samples if the
writer prepared some for us.

Fixes #545807
2009-08-24 13:30:11 +02:00
Tim-Philipp Müller
0021e6b765 Revert inlines that cause compiler warnings and are not needed anyway 2009-08-08 17:51:10 +01:00
Edward Hervey
9329b8be72 gst-libs: Remove dead assignments and resulting unused variables. 2009-08-08 15:54:57 +02:00
Wim Taymans
090808a295 baseaudiosrc: change default slave method
Set the default slave method to the much better skew slaving algortihm.
2009-08-06 12:58:58 +02:00
Olivier Crête
429d3555a2 audiofilter: Don't assert on slightly different caps
Plugins should not assert on incompatible caps, caps negotiation will
fail anyway.
2009-07-30 14:34:05 +01:00
Olivier Crête
4e88633de4 audiosink: Add stream-status messages
Fixes #587695
2009-07-20 12:54:37 +02:00
Olivier Crête
cc0da016f8 audiosrc: Add stream-status messages
See #587695
2009-07-20 12:54:37 +02:00
Stefan Kost
0e967f1b14 multichannel: rewrite the new doc comment a bit
Its part of the audio lib.
2009-06-29 17:49:58 +03:00
Wim Taymans
8601862e27 ringbuffer: add vmethod to clear the ringbuffer
Add a vmethod so that subclasses can be notified when they should clear the data
in the ringbuffer.
2009-06-29 15:17:25 +02:00
Stefan Kost
57a7d6f699 docs: add basic section docs for multichannel and relocate the ones for audio
Add section docs for multichannel, so that it has a short desc in the toc too.
Move the section docs in adio up, so that the follow the copyright like
elsewhere.
2009-06-27 23:25:09 +03:00
Wim Taymans
ffd90dda89 audiosrc: fix get_offset
When we need to jump to the most recently captured sample, jump to where the
next sample will be written instead of to some old data.

Fixes #581460
2009-06-17 14:00:23 +02:00
Wim Taymans
57a13f28de audiosink: free the ringbuffer when going to NULL
Unparent and free the ringbuffer when going to NULL, like we do with the
audiosrc element. We can do this now because we correctly manage the time
jumping back to 0.
2009-06-17 13:18:18 +02:00
Wim Taymans
e4492c24ea audio: correctly handle short read/writes 2009-06-17 13:17:30 +02:00
René Stadler
2c5f455423 baseaudiosrc: add some extra logging for buffer timestamps 2009-06-17 12:36:50 +02:00
Tim-Philipp Müller
70089160f8 audiosink, audiosrc: do the class_ref()s in the right class_init functions
Spotted by Philip Jägenstedt. Hopefully fixes #585970 for real.
2009-06-16 14:14:26 +01:00
Tim-Philipp Müller
3767cb6005 audiosink,audiosrc: ref the audio ring buffer class and type in class_init
Hack around thread-safety issues in GObject and our racy _get_type()
functions (we could easily fix the _get_type() functions, but we still
need to hack around the GObject class races until we require a newer
GLib version, I think).
2009-06-15 15:39:09 +01:00
Wim Taymans
a5491ba218 audiosrc: return FALSE when receiving a SEEK event
When receiving a seek event, return FALSE as we don't implement seeking.
2009-06-15 12:57:39 +02:00
Wim Taymans
a9c82f9472 ringbuffer: handle border cases in resampler 2009-06-11 19:13:28 +02:00
Wim Taymans
69b7fb3845 baseaudiosink: reset accum when dropping samples
When we are resampling and we drop samples because we paused, reset the accum
counter because it's now invalid.
2009-06-11 12:38:35 +02:00
Tim-Philipp Müller
249d9b4aa1 Don't include config.h multiple times when build audio testchannel app.
Fixes build problem on win32 (#585075).
2009-06-10 21:37:29 +01:00
Wim Taymans
38e59ec75d baseaudiosink: no need to cause discont when clipping
Remove the discont-when-clipping hack now that basesink provides us with
correctly clipped samples when stepping.
2009-06-09 12:09:15 +02:00
Wim Taymans
cb4952fc2e audiosink: don't align when we clip
Don't align samples when they were clipped. Not entirely correct but better than
nothing for now.
2009-06-08 17:26:59 +02:00
Andy Wingo
c7ca6abe53 add can-activate-pull property to baseaudiosink
* gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
  to baseaudiosink.
2009-05-26 13:17:44 +02:00
Wim Taymans
81170c4989 audiosink: improve debug message 2009-05-21 10:48:49 +02:00
Wim Taymans
c68a361e31 audiosink: return the return value of wait_preroll
Return the value that _wait_preroll() returned instead of always WRONG_STATE.
2009-05-19 17:17:37 +02:00
Wim Taymans
b9723f6e1c audioclock: make our internal time monotonic
Make the internal time increase monotonically.
2009-05-13 21:38:56 +02:00
Wim Taymans
d655120ee6 audioclock: make sure values are ever increasing 2009-05-12 10:39:41 +02:00
Andy Wingo
9f74ce745f Revert "add can-activate-pull property to baseaudiosink"
This reverts commit c4074a2ee4.
2009-04-29 11:18:42 +02:00
Andy Wingo
219a31fa3c Revert "[baseaudiosink] add docs for can-activate-pull"
This reverts commit 416ce16f26.
2009-04-29 11:18:33 +02:00
Andy Wingo
416ce16f26 [baseaudiosink] add docs for can-activate-pull
* gst-libs/gst/audio/gstbaseaudiosink.c: Add documentation for
  can-activate-pull.
2009-04-28 18:48:33 +02:00
Andy Wingo
c4074a2ee4 add can-activate-pull property to baseaudiosink
* gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
  to baseaudiosink.
2009-04-28 18:28:50 +02:00
Wim Taymans
32904de58f baseaudiosink: don't unparent the ringbuffer
when going to NULL, don't unparent the ringbuffer because we don't support going
back to 0 very well yet.
Fixes #579203
2009-04-17 11:03:32 +02:00
Stefan Kost
ab24d9d65c log: use G_GUINT64_FORMAT instead of llu 2009-04-15 00:02:39 +03:00
Wim Taymans
dffd1bcc97 baseaudiosrc: adjust the internal timestamp
Adjust the internal timestamp before comparing it against the adjusted clock
time.
Fixes #578506
2009-04-14 13:16:14 +02:00
Wim Taymans
0c4c1410f9 baseaudiosink: use new clock time methods
Use the unadjusted internal clock times to calculate the internal/external
offset when calibrating the clock.

When going to NULL, unparent and free the ringbuffer, like we do in the source
element.
See #578506
2009-04-14 13:12:59 +02:00
Wim Taymans
4231d54823 audioclock: add methods for the internal offset
Add two methods for getting the unadjusted time of the clock and one for
adjusting an internal time. We will need these methods for correctly handling
the time after a gst_audio_clock_reset().

Add a debug category and some debug lines to the audio clock.

API: gst_audio_clock_get_time()
API: gst_audio_clock_adjust()
API: GST_AUDIO_CLOCK_CAST()
2009-04-14 13:08:52 +02:00
Wim Taymans
251f152c20 baseaudiosink: use the internal clock time
We can't assume that the internal clock time is the same as the function we
installed on our provided clock because somebody might have changed it.
2009-04-10 21:50:55 +02:00
Wim Taymans
e6798c5cce ringbuffer: allow for custom commit functions
Allow subclasses to override the commit method.
2009-04-09 18:04:44 +02:00
Wim Taymans
cae2981f83 baseaudiosink: fix a small glitch after pause
After we pause the stream and interrupt the writeout to the ringbuffer, also adjust
the amount of output samples we consumed. We can't do this reliably with the
current API when we are doing trick modes but we can do the right thing for
normal playback.
2009-04-08 18:06:54 +02:00
Stefan Kost
ff9ee1dc5a audiofilter: don't leak pad-template
gst_element_class_add_pad_template() does not take ownership.
2009-04-07 22:39:07 +03:00
Tim-Philipp Müller
0267e79778 audiosrc: improve 'Dropped n samples' warning message 2009-03-25 11:27:44 +00:00
Stefan Kost
251e4d160a docs: don't put random stuff in tags.
Tags like Since: or Returns: can only be followed by more tags. gtk-doc has no
tag to append text again to the documentation body.
2009-02-26 10:09:59 +02:00
Stefan Kost
486fe43cb9 Add a FIXME 0.11. Make the log message a bit more detailed and add comments. 2009-02-02 18:05:42 +02:00
Stefan Kost
950d0c0a7d Link to the class, as we can't link to the members yet. 2009-01-31 18:44:32 +02:00
Jan Schmidt
63c9ede3d0 Extend and clean up git ignores 2009-01-23 23:16:11 +00:00
José Alburquerque
7431789249 gst-libs/gst/audio/gstaudioclock.*: Make gst_audio_clock_new use const gchar* to ease the wrapping of
Original commit message from CVS:
Patch by: José Alburquerque <jaalburqu svn gnome org>
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_new):
* gst-libs/gst/audio/gstaudioclock.h:
Make gst_audio_clock_new use const gchar* to ease the wrapping of
C++ bindings. Fixes #566723.
2009-01-06 17:30:31 +00:00
Wim Taymans
0a4c1bc64c gst-libs/gst/audio/gstbaseaudiosink.c: Avoid holding the OBJECT_LOCK when calling ringbuffer functions that take the ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_change_state):
Avoid holding the OBJECT_LOCK when calling ringbuffer functions that
take the ringbuffer lock because rinbuffer lock > OBJECT_LOCK. We can do
this because the async_play method is deprecated and usually not called
anymore.
2009-01-05 17:13:13 +00:00
Edward Hervey
e2fcc71650 Switch glib-mkenum for gst-libs/gst/audio from multichannel- to audio- in order to wrap all enums declarations of tha...
Original commit message from CVS:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/multichannel.h:
* gst-libs/gst/audio/testchannels.c:
* win32/MANIFEST:
* win32/common/audio-enumtypes.c:
(gst_audio_channel_position_get_type),
(gst_ring_buffer_state_get_type),
(gst_ring_buffer_seg_state_get_type),
(gst_buffer_format_type_get_type), (gst_buffer_format_get_type):
* win32/common/audio-enumtypes.h:
* win32/common/multichannel-enumtypes.c:
* win32/common/multichannel-enumtypes.h:
* win32/vs6/grammar.dsp:
* win32/vs6/libgstaudio.dsp:
* win32/vs7/libgstaudio.vcproj:
* win32/vs8/libgstaudio.vcproj:
Switch glib-mkenum for gst-libs/gst/audio from multichannel- to
audio- in order to wrap all enums declarations of that library.
This modification should not matter since that header file is not a
public header (it will be included by public headers).
Modify win32 crap^Wfiles accordingly.
2008-12-31 11:20:26 +00:00
Edward Hervey
20adaa1328 gst-libs/gst/audio/: Complete Sebastien's commit from the 13th by exporting the _slave_method_get_type() methods.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstbaseaudiosink.h:
Complete Sebastien's commit from the 13th by exporting the
_slave_method_get_type() methods.
2008-12-30 17:55:07 +00:00
Wim Taymans
a579eba73d gst-libs/gst/audio/gstbaseaudiosink.c: Pause the write thread before deactivating and releasing the ringbuffer to avo...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_setcaps):
Pause the write thread before deactivating and releasing the ringbuffer
to avoid a deadlock when we do gapless playback with different sample
rates in playbin2.  Fixes #564929.
2008-12-20 12:45:03 +00:00
Sebastian Dröge
4ed1f5d6fd gst-libs/gst/audio/gstbaseaudiosrc.c: Make GstAudioSrcSlaveMethod get_type() function non-static as it's public now.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
Make GstAudioSrcSlaveMethod get_type() function non-static
as it's public now.
* win32/common/libgstaudio.def:
* win32/common/libgstnetbuffer.def:
Add some missing functions to the list of exported symbols.
2008-12-19 13:03:00 +00:00
Sebastian Dröge
04d9ff9a24 gst-libs/gst/audio/: API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API. This is needed for the C...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_slave_method_get_type),
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_slave_method_get_type),
(gst_base_audio_src_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the
public API. This is needed for the C++ bindings to be able
to use this base classes. Fixes bug #564200, #564206.
2008-12-13 06:57:09 +00:00
Wim Taymans
af354dbef3 gst-libs/gst/audio/gstbaseaudiosrc.c: Avoid nasty int overflows after about 12 hours and 25 minutes when these code p...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_get_offset), (gst_base_audio_src_create):
Avoid nasty int overflows after about 12 hours and 25 minutes when these
code paths are triggered.
A free beer to Håvard Graff for finding this!
2008-11-27 16:47:41 +00:00
Wim Taymans
6983c1c85b gst-libs/gst/audio/gstbaseaudiosink.c: Really fix audiosink drain handling by keeping track of the running_time of th...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_event),
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
Really fix audiosink drain handling by keeping track of the running_time
of the last sample.
2008-11-25 10:32:49 +00:00
Stefan Kost
a8264f66c7 gst-libs/gst/audio/gstbaseaudiosink.c: Time is already in running_time. Remove base_time handling. Fixes audiosinks n...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
Time is already in running_time. Remove base_time handling. Fixes
audiosinks not draining and thus chopping some audio in the end.
2008-11-24 20:11:52 +00:00
Stefan Kost
7f937c99d4 gst-libs/gst/audio/gstbaseaudiosink.c: Add one log message to check for audio_drained. Sync one log message with the ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
Add one log message to check for audio_drained. Sync one log message
with the condition. Send EOS after draining audio in pull mode.
2008-11-24 12:56:54 +00:00
Wim Taymans
e701e64005 gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting for the ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
(gst_base_audio_sink_callback):
Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting
for the latency to expire, fixes #559567.
2008-11-10 14:22:09 +00:00
Wim Taymans
6eed8ca285 gst-libs/gst/audio/gstaudiosink.c: Implement a separate activate functions to start monitoring the segments or, in pu...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_class_init), (gst_audioringbuffer_acquire),
(gst_audioringbuffer_activate), (gst_audioringbuffer_release),
(gst_audioringbuffer_stop):
Implement a separate activate functions to start monitoring the segments
or, in pull mode, pulling in data.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_init), (gst_base_audio_sink_dispose),
(gst_base_audio_sink_query_pad), (gst_base_audio_sink_query),
(gst_base_audio_sink_setcaps), (gst_base_audio_sink_callback),
(gst_base_audio_sink_activate_pull),
(gst_base_audio_sink_async_play),
(gst_base_audio_sink_change_state):
Implement pad and element convert query function.
Activate the ringbuffer.
Use the segment last_stop value as the offset to pull.
Use new basesink _do_preroll() method to preroll in the pulling thread.
Take appropriate locking in the pulling thread.
* gst-libs/gst/audio/gstringbuffer.h:
Update some docs.
2008-10-20 15:35:37 +00:00
Wim Taymans
a6b78893c0 Add methods to more accuratly control the pulling thread of a ringbuffer.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_convert),
(gst_ring_buffer_activate), (gst_ring_buffer_is_active):
* gst-libs/gst/audio/gstringbuffer.h:
Add methods to more accuratly control the pulling thread of a
ringbuffer.
Add format conversion helper code to the ringbuffer.
API: GstRingBuffer:gst_ring_buffer_activate()
API: GstRingBuffer:gst_ring_buffer_is_active()
API: GstRingBuffer:gst_ring_buffer_convert()
2008-10-17 13:19:05 +00:00
Wim Taymans
927999603a gst-libs/gst/audio/gstaudiosink.c: Signal thread startup earlier so that we can immediatly go into pull mode when we ...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
(gst_audioringbuffer_stop):
Signal thread startup earlier so that we can immediatly go into pull
mode when we have to and block on preroll.
2008-10-16 15:44:37 +00:00
Wim Taymans
7bd29abb9d gst-libs/gst/audio/gstringbuffer.c: In pull mode we want the callback to prepull a buffer we can preroll on even when...
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_prepare_read):
In pull mode we want the callback to prepull a buffer we can preroll on
even when we are not yet playing.
2008-10-16 15:38:50 +00:00
Edward Hervey
57b0f5bef6 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix debug statements (space between '%' and actual format).
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Fix debug statements (space between '%' and actual format).
2008-10-08 15:30:33 +00:00
Håvard Graff
11086cf6f8 gst-libs/gst/audio/gstbaseaudiosrc.c: Implement skew clock slaving. Fixes #552559.
Original commit message from CVS:
Patch by: Håvard Graff <havard dot graff at tandberg dot com>
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Implement skew clock slaving. Fixes #552559.
2008-10-08 09:12:36 +00:00
Wim Taymans
dd01a1e56a gst-libs/gst/audio/: Fix include of config.h
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
* gst-libs/gst/audio/testchannels.c:
Fix include of config.h
2008-10-08 09:10:23 +00:00
Tim-Philipp Müller
b579580991 gst-libs/gst/audio/audio.h: Remove trailing comma from enum list, which causes problems with -pendantic (#550729).
Original commit message from CVS:
* gst-libs/gst/audio/audio.h: (GST_AUDIO_FIELD_SIGNED):
Remove trailing comma from enum list, which causes problems
with -pendantic (#550729).
2008-09-13 11:04:02 +00:00
Wim Taymans
265a494de5 gst-libs/gst/audio/gstaudiosrc.c: Disable a code path that is now called but causes a deadlock for some reason and is...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_stop):
Disable a code path that is now called but causes a deadlock for some
reason and is unneeded.
2008-09-04 16:25:06 +00:00
Wim Taymans
da76d5e7cb gst-libs/gst/audio/gstaudiosink.c: Since we now call stop, we trigger this code path that causes a deadlock is appare...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_stop):
Since we now call stop, we trigger this code path that causes a deadlock
is apparently not needed.
2008-08-26 17:24:31 +00:00
Wim Taymans
440432612b gst-libs/gst/audio/gstringbuffer.c: Also allow the case where the ringbuffer was paused when we try to stop it so tha...
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
(gst_ring_buffer_stop):
Also allow the case where the ringbuffer was paused when we try to stop
it so that the basesrc stop function is still called.
2008-08-26 15:45:36 +00:00
Wim Taymans
510a5befc1 gst-libs/gst/audio/gstbaseaudiosrc.c: When not slaved to another clock also subtract the base_time from our internal ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
When not slaved to another clock also subtract the base_time from our
internal clock time to get the running time.
2008-08-13 09:17:38 +00:00
Stefan Kost
5d2049cdb3 gst-libs/gst/audio/Makefile.am: Don't try to build that example anymore.
Original commit message from CVS:
* gst-libs/gst/audio/Makefile.am:
Don't try to build that example anymore.
2008-08-11 15:05:35 +00:00
Stefan Kost
3511b2772b gst-libs/gst/audio/: Move audiofiltertemplate to gst-template.
Original commit message from CVS:
* gst-libs/gst/audio/.cvsignore:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
* gst-libs/gst/audio/make_filter:
Move audiofiltertemplate to gst-template.
2008-08-11 14:51:58 +00:00
Stefan Kost
01554ac056 More docs and shuffling. What can we do with the hundreds of #defines.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudiosrc.h:
More docs and shuffling. What can we do with the hundreds of #defines.
2008-08-11 09:20:33 +00:00
Stefan Kost
f73aa5b817 gst-libs/gst/: Reducing number of dundocumented symbols.
Original commit message from CVS:
* gst-libs/gst/audio/audio.h:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/audio/gstringbuffer.h:
* gst-libs/gst/interfaces/propertyprobe.h:
* gst-libs/gst/tag/gsttagdemux.h:
Reducing number of dundocumented symbols.
2008-08-11 08:34:56 +00:00
Stefan Kost
26ad0ba982 gst-libs/gst/audio/audio.c: Fix doc comment syntax.
Original commit message from CVS:
* gst-libs/gst/audio/audio.c:
Fix doc comment syntax.
* gst-libs/gst/interfaces/propertyprobe.c:
Add more doc-comments and a FIXME: for the signal.
2008-08-11 07:16:30 +00:00
Frederic Crozat
89be246154 Make sure gettext returns translations in UTF-8 encoding rather than in the current locale encoding (#546822).
Original commit message from CVS:
Patch by: Frederic Crozat <fcrozat@mandriva.org>
* ext/alsa/gstalsaplugin.c: (plugin_init):
* ext/cdparanoia/gstcdparanoiasrc.c: (plugin_init):
* ext/gnomevfs/gstgnomevfs.c: (plugin_init):
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init):
* gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
* gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal):
* gst/playback/gstdecodebin.c: (plugin_init):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_plugin_init):
* gst/playback/gstplayback.c: (plugin_init):
* gst/playback/gstqueue2.c: (plugin_init):
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_plugin_init):
* sys/v4l/gstv4l.c: (plugin_init):
Make sure gettext returns translations in UTF-8 encoding rather
than in the current locale encoding (#546822).
2008-08-07 15:58:58 +00:00
Wim Taymans
d2f328f55b gst-libs/gst/audio/gstbaseaudiosink.c: Report latency even if we are not live instead of hiding it.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_query), (gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_render):
Report latency even if we are not live instead of hiding it.
Take ts-offset and render-delay of the basesink into account when
scheduling samples.
Rework the clipping code so that we can take the various offsets into
account and still do correct clipping.
2008-06-20 09:09:37 +00:00
Sebastian Dröge
0de81029c8 API: Make gst_audio_check_channel_positions() public.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions):
* gst-libs/gst/audio/multichannel.h:
API: Make gst_audio_check_channel_positions() public.
* tests/check/libs/audio.c: (GST_START_TEST):
Add some simple checks for gst_audio_check_channel_positions().
2008-06-03 08:48:32 +00:00
Mark Nauwelaerts
9fa61c528d gst-libs/gst/audio/gstbaseaudiosrc.c: Add a gtk-doc chunk for the new properties to have a Since: indication.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
Add a gtk-doc chunk for the new properties to have a Since: indication.
2008-05-31 19:57:57 +00:00
Mark Nauwelaerts
c660bbd6dd gst-libs/gst/audio/gstbaseaudiosrc.c: Provide readable actual-buffer-time and actual-latency-time properties that ref...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
(gst_base_audio_src_get_property), (gst_base_audio_src_setcaps),
(gst_base_audio_src_change_state):
Provide readable actual-buffer-time and actual-latency-time properties
that reflect the configured ringbuffer values. Fixes #524724.
2008-05-31 18:10:47 +00:00
Sebastian Dröge
45ef6b5e13 gst-libs/gst/audio/multichannel.c: Allow rear center together with rear left/right and other previously conflicting c...
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(gst_audio_fixate_channel_positions):
Allow rear center together with rear left/right and other previously
conflicting channel positions. The reason why they weren't allowed
was the channel mixing implementation in audioconvert.
Also take this into account when fixing channel layouts.
Allow setting channel positions for 1/2 channels when using
gst_audio_set_structure_channel_position().
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_compatible), (gst_channel_mix_detect_pos),
(gst_channel_mix_fill_one_other), (gst_channel_mix_fill_others),
(gst_channel_mix_fill_special), (gst_channel_mix_fill_matrix):
Major rewrite of the channel mixing.
We now allow previously	conflicting channel positions to appear
together (rear center and rear left/right for example).
Fixes bug #533817.
Rework the way channels are mixed together to take more possible
channel positions into account, properly mix from/to side channels
and don't assume that either center, left&right or nothing of a
specific position is available anymore.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Adjust unit tests with non-standard 1/2 channel layouts to the more
correct new behaviour.
Add a unit test for 5.1->Stereo downmixing.
2008-05-29 11:34:09 +00:00
Wim Taymans
35e4b75b86 gst-libs/gst/audio/gstaudioclock.*: Add method to inform the clock that the time starts from 0 again. We use this inf...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
(gst_audio_clock_reset), (gst_audio_clock_get_internal_time):
* gst-libs/gst/audio/gstaudioclock.h:
Add method to inform the clock that the time starts from 0 again. We use
this info to calculate a clock offset so that the time we report in
internal_time is monotonically increasing, as required by the clock base
class. Fixes #521761.
API: GstAudioClock::gst_audio_clock_reset()
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Reset reported time when we (re)create the ringbuffer.
2008-05-27 16:20:17 +00:00
Sebastian Dröge
d03bbd1e3e gst-libs/gst/audio/multichannel.c: Allow non-standard 2 channel layouts.
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions):
Allow non-standard 2 channel layouts.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add some tests for converting and remapping non-standard 1 and 2
channel layouts.
2008-05-21 07:39:56 +00:00
Wim Taymans
f36d9d6b08 gst-libs/gst/audio/gstbaseaudiosink.c: We can only use our optimal calibration if we prerolled before the latency exp...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_sync_latency):
We can only use our optimal calibration if we prerolled before the
latency expired.
2008-05-20 16:26:53 +00:00
Wim Taymans
d8dc371c0d ext/gnomevfs/gstgnomevfssrc.*: Set the ICY caps on the srcpad from where they get picked up by the base class now and...
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
(gst_gnome_vfs_src_finalize),
(gst_gnome_vfs_src_received_headers_callback),
(gst_gnome_vfs_src_create), (gst_gnome_vfs_src_stop):
* ext/gnomevfs/gstgnomevfssrc.h:
Set the ICY caps on the srcpad from where they get picked up by the base
class now and set on the outgoing buffers.
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
BaseSrc now sets the caps on outgoing buffers automatically.
2008-05-20 11:13:27 +00:00
Wim Taymans
95d162fb71 gst-libs/gst/audio/gstbaseaudiosink.c: Change the way in which the ringbuffer is started when dealing with a slaved c...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_resample_slaving),
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
(gst_base_audio_sink_async_play),
(gst_base_audio_sink_change_state):
Change the way in which the ringbuffer is started when dealing with a
slaved clock and latency. We now sync to the clock until we reach
upstream latency before starting the ringbuffer. This has the effect
that we can accurately align the master and slave clocks and let the
rate correction code take care of the initial drift or rounding errors
instead of leaving them uncorrected with the old approach.
2008-05-20 11:09:06 +00:00
Wim Taymans
0c9b13988c gst-libs/gst/audio/gstbaseaudiosink.c: Revert previous patch that attempted to more accurately calculate the initial ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
Revert previous patch that attempted to more accurately calculate the
initial offset between master and slave clock. The best thing we can do
in general is take the time of both clocks as the diff since we don't
know when the actual preroll happened.
2008-05-12 08:45:11 +00:00
Wim Taymans
fc523e047c gst-libs/gst/audio/gstaudiosink.c: Choose to allocate one less segment but require one additional segment as latency.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
Choose to allocate one less segment but require one additional segment
as latency.
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_acquire):
No need to increment the number of segments in the source.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_time), (clock_convert_external),
(gst_base_audio_sink_resample_slaving),
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_none_slaving), (gst_base_audio_sink_render),
(gst_base_audio_sink_async_play):
Remove adding latency when returning the internal time while subtracting
it again when we use the value a little later.
When calculating the end timestamp, we are making a rounding error
with the current algorithm. Ensure that we don't accumulate these
rounding errors when aligning samples by not resampling at all if we
don't need to. Fixes #419351.
Make the initial calibration of the clock slaving a little more
predictable and accurate. Also handle the case where we don't do
clock slaving.
2008-05-09 16:38:10 +00:00
Wim Taymans
09f7dee84d gst-libs/gst/audio/gstbaseaudiosink.c: Report the latency with the new seglatency parameter.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_query):
Report the latency with the new seglatency parameter.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_debug_spec_buff), (gst_ring_buffer_parse_caps),
(gst_ring_buffer_acquire):
* gst-libs/gst/audio/gstringbuffer.h:
Add new field to the ringbufferspec to specify the expected latency
between the underlying device read/write pointer, this is needed
when writing sinks that sit a little closer to the hardware.
Add some more docs for other fields.
2008-05-07 15:47:03 +00:00
Sebastian Dröge
83f0729394 Remove some unused code.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
* gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
* gst/tcp/gsttcp.c: (gst_tcp_socket_write):
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_get_fps_list):
Remove some unused code.
* gst/audioconvert/gstaudioquantize.c:
(gst_audio_quantize_free_noise_shaping):
Don't return before freeing the noise shaping history.
2008-05-04 15:02:20 +00:00
Wim Taymans
7916e386ca gst-libs/gst/audio/gstbaseaudiosink.h: Clarify some docs.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.h:
Clarify some docs.
* gst-libs/gst/audio/gstbaseaudiosrc.c: (slave_method_get_type),
(gst_base_audio_src_class_init), (gst_base_audio_src_init),
(gst_base_audio_src_set_slave_method),
(gst_base_audio_src_get_slave_method),
(gst_base_audio_src_set_property),
(gst_base_audio_src_get_property), (gst_base_audio_src_create):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
Add property and methods for selecting the clock slave method in the
source, like in the sink.
We only implement "none" and "re-timestamp" for now.
API: gst_base_audio_src_set_slave_method()
API: gst_base_audio_src_get_slave_method()
2008-04-28 08:51:38 +00:00
Sebastian Dröge
66bbadadd0 gst-libs/gst/audio/gstringbuffer.c: Use g_atomic_int_set() instead of gst_atomic_int_set().
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_may_start):
Use g_atomic_int_set() instead of gst_atomic_int_set().
2008-04-17 07:33:46 +00:00
Tim-Philipp Müller
7a29d716bd gst/: Work around missing bits of thread-safety on older GLibs some more to avoid assertions when starting up multipl...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
* gst/playback/gstplayback.c: (plugin_init):
* gst/volume/gstvolume.c: (plugin_init):
Work around missing bits of thread-safety on older GLibs some
more to avoid assertions when starting up multiple playbin
objects concurrently (see #512382).
2008-04-06 20:16:27 +00:00
Wim Taymans
ce67ac6373 gst-libs/gst/audio/gstbaseaudiosink.c: Guard against over and underflows because of clock slaving.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render):
Guard against over and underflows because of clock slaving.
When we are using our own clock, still compensate for any calibrations
that we might have done to our clock.
2008-04-03 10:37:03 +00:00
Wim Taymans
877a45b791 gst-libs/gst/audio/gstaudiosink.c: Small debug improvement.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_release):
Small debug improvement.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Fix bug in determining the sample start/stop position, we want to base
this decision on the fact that we are going forwards or backwards, not
slower or faster. This fixes some ugly resync warnings when playing at
very slow speeds.
2008-03-24 11:24:22 +00:00
Sebastian Dröge
49deb0c05d Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u...
Original commit message from CVS:
* configure.ac:
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_class_init):
* ext/alsa/gstalsasink.c: (gst_alsasink_class_init):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
* ext/cdparanoia/gstcdparanoiasrc.c:
(gst_cd_paranoia_src_class_init):
* ext/gio/gstgiosink.c: (gst_gio_sink_class_init):
* ext/gio/gstgiosrc.c: (gst_gio_src_class_init):
* ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init):
* ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init):
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
* ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
* ext/pango/gsttextrender.c: (gst_text_render_class_init):
* ext/theora/theoradec.c: (gst_theora_dec_class_init):
* ext/theora/theoraenc.c: (gst_theora_enc_class_init):
* ext/theora/theoraparse.c: (gst_theora_parse_class_init):
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init):
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_class_init):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_class_init):
* gst-libs/gst/interfaces/mixertrack.c:
(gst_mixer_track_class_init):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init):
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init):
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init):
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_class_init):
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init):
* gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(preroll_unlinked):
* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init):
* gst/playback/gstplaysink.c: (gst_play_sink_class_init):
* gst/playback/gstqueue2.c: (gst_queue_class_init):
* gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
* gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
(gst_stream_selector_class_init):
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
* gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
* gst/videorate/gstvideorate.c: (gst_video_rate_class_init):
* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_video_test_src_class_init):
* gst/volume/gstvolume.c: (gst_volume_class_init):
* sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init):
* sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
* sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init):
* sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use
static strings (i.e. all). This gives us less memory usage,
fewer allocations and thus less memory defragmentation. Depend
on core CVS for this. Fixes bug #523806.
2008-03-22 15:00:53 +00:00
Michael Smith
15e209d20e gst-libs/gst/audio/gstringbuffer.*: Rename recently added buffer types to make more sense.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
* gst-libs/gst/audio/gstringbuffer.h:
Rename recently added buffer types to make more sense.
* ext/alsa/gstalsasink.c: (alsasink_parse_spec),
(gst_alsasink_write):
Adapt for above API changes.
Fixes bug #520523.
2008-03-12 12:39:13 +00:00
Wim Taymans
579949e2c5 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix duration when no clock was provided. Fixes #520300.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Fix duration when no clock was provided. Fixes #520300.
2008-03-10 17:19:56 +00:00
Sebastian Dröge
ec7afb6f84 Correct all relevant warnings found by the sparse semantic code analyzer. This include marking several symbols static...
Original commit message from CVS:
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_get_type):
* ext/alsa/gstalsasink.c: (set_hwparams):
* ext/alsa/gstalsasrc.c: (set_hwparams):
* ext/gio/gstgio.c: (gst_gio_uri_handler_get_uri):
* ext/ogg/gstoggmux.h:
* ext/ogg/gstogmparse.c:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/fft/kiss_fft_f64.c: (kiss_fft_f64_alloc):
* gst-libs/gst/pbutils/missing-plugins.c:
(gst_missing_uri_sink_message_new),
(gst_missing_element_message_new),
(gst_missing_decoder_message_new),
(gst_missing_encoder_message_new):
* gst-libs/gst/rtp/gstbasertppayload.c:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
(gst_rtcp_packet_bye_get_reason):
* gst/audioconvert/gstaudioconvert.c:
* gst/audioresample/gstaudioresample.c:
* gst/ffmpegcolorspace/imgconvert.c:
* gst/playback/test.c: (gen_video_element), (gen_audio_element):
* gst/typefind/gsttypefindfunctions.c:
* gst/videoscale/vs_4tap.c:
* gst/videoscale/vs_4tap.h:
* sys/v4l/gstv4lelement.c:
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_any_caps):
* sys/v4l/v4l_calls.c:
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_init),
(gst_v4lsrc_try_capture):
* sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
(gst_ximagesink_ximage_new):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
(gst_xvimagesink_xvimage_new):
* tests/check/elements/audioconvert.c:
* tests/check/elements/audioresample.c:
(fail_unless_perfect_stream):
* tests/check/elements/audiotestsrc.c: (setup_audiotestsrc):
* tests/check/elements/decodebin.c:
* tests/check/elements/gdpdepay.c: (setup_gdpdepay),
(setup_gdpdepay_streamheader):
* tests/check/elements/gdppay.c: (setup_gdppay), (GST_START_TEST),
(setup_gdppay_streamheader):
* tests/check/elements/gnomevfssink.c: (setup_gnomevfssink):
* tests/check/elements/multifdsink.c: (setup_multifdsink):
* tests/check/elements/textoverlay.c:
* tests/check/elements/videorate.c: (setup_videorate):
* tests/check/elements/videotestsrc.c: (setup_videotestsrc):
* tests/check/elements/volume.c: (setup_volume):
* tests/check/elements/vorbisdec.c: (setup_vorbisdec):
* tests/check/elements/vorbistag.c:
* tests/check/generic/clock-selection.c:
* tests/check/generic/states.c: (setup), (teardown):
* tests/check/libs/cddabasesrc.c:
* tests/check/libs/video.c:
* tests/check/pipelines/gio.c:
* tests/check/pipelines/oggmux.c:
* tests/check/pipelines/simple-launch-lines.c:
(simple_launch_lines_suite):
* tests/check/pipelines/streamheader.c:
* tests/check/pipelines/theoraenc.c:
* tests/check/pipelines/vorbisdec.c:
* tests/check/pipelines/vorbisenc.c:
* tests/examples/seek/scrubby.c:
* tests/examples/seek/seek.c: (query_positions_elems),
(query_positions_pads):
* tests/icles/stress-xoverlay.c: (myclock):
Correct all relevant warnings found by the sparse semantic code
analyzer. This include marking several symbols static, using
NULL instead of 0 for pointers and using "foo (void)" instead
of "foo ()" for declarations.
* win32/common/libgstrtp.def:
Add gst_rtp_buffer_set_extension_data to the symbol definition file.
2008-03-03 06:04:31 +00:00
Julien Moutte
f0154849b0 ext/alsa/gstalsa.c: Probe for IEC958 pcm to detect if we can do SPDIF output.
Original commit message from CVS:
2008-02-29  Julien Moutte  <julien@fluendo.com>

* ext/alsa/gstalsa.c: (gst_alsa_open_iec958_pcm),
(gst_alsa_probe_supported_formats): Probe for IEC958 pcm to
detect
if we can do SPDIF output.
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec),
(gst_alsasink_prepare), (gst_alsasink_close),
(gst_alsasink_write):
* ext/alsa/gstalsasink.h: Initial support for SPDIF.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_parse_caps):
* gst-libs/gst/audio/gstringbuffer.h: Add non linear buffer
types
to support AC3, EC3 and IEC958 buffers.
2008-02-29 18:44:36 +00:00
Tim-Philipp Müller
2c538ea740 gst-libs/gst/audio/multichannel.c: Fix confusing terminology in docs and code: structure fields are 'fields' and not ...
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(GST_AUDIO_CHANNEL_POSITIONS_FIELD_NAME),
(gst_audio_get_channel_positions), (gst_audio_set_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(add_list_to_struct), (gst_audio_set_caps_channel_positions_list),
(gst_audio_fixate_channel_positions):
Fix confusing terminology in docs and code: structure fields are
'fields' and not 'properties'.
2008-02-19 20:42:09 +00:00
Tim-Philipp Müller
a1e59086ba gst-libs/gst/audio/multichannel.c: Give more useful warning messages if one of the channel layout enums passed to us ...
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions), (add_list_to_struct):
Give more useful warning messages if one of the channel
layout enums passed to us is invalid and if the "channels"
field in the caps has a GType we don't expect.
2008-02-19 20:36:58 +00:00
Tim-Philipp Müller
29162d0a46 gst-libs/gst/audio/multichannel.c: Fix typo in docs blurb.
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
Fix typo in docs blurb.
2008-02-19 20:22:09 +00:00
Sebastian Dröge
a6e4222c70 gst-libs/gst/audio/gstaudiofilter.c: Initialize the GstRingerBuffer class to get it's debug category initialized. gst...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_class_init):
Initialize the GstRingerBuffer class to get it's debug category
initialized. gst_ring_buffer_parse_caps() uses the ringbuffer debug
category and otherwise we get some g_critical(). Fixes bug #512334.
2008-01-29 09:47:12 +00:00
Tim-Philipp Müller
3feb4bc8c5 gst-libs/gst/audio/: Ref audio clock class from a thread-safe context to make sure however unlikely that may be in pr...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
Ref audio clock class from a thread-safe context to make sure
we're not bit by GObjects lack of thread-safety here (#349410),
however unlikely that may be in practice.
2008-01-10 17:55:53 +00:00
Sebastian Dröge
a000758477 gst-libs/gst/audio/gstaudiofilter.c: Don't set element details for the abstract GstAudioFilter class.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type):
Don't set element details for the abstract GstAudioFilter class.
2008-01-03 07:17:05 +00:00
Sebastian Dröge
0e5857ea26 gst-libs/gst/audio/gstaudiofilter.c: Implement get_unit_size() vmethod of GstBaseTransform.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_class_init), (gst_audio_filter_get_unit_size):
Implement get_unit_size() vmethod of GstBaseTransform.
2008-01-02 12:09:48 +00:00
Wim Taymans
355e8a940d gst-libs/gst/audio/gstaudiosink.c: Improve debug output.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
(gst_audioringbuffer_open_device),
(gst_audioringbuffer_close_device), (gst_audioringbuffer_acquire),
(gst_audioringbuffer_release), (gst_audioringbuffer_start),
(gst_audioringbuffer_pause), (gst_audioringbuffer_stop),
(gst_audio_sink_create_ringbuffer):
Improve debug output.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
(gst_ring_buffer_pause), (gst_ring_buffer_delay):
Prevent some functions from doing things and failing when the
ringbuffer is not yet acquired.
2007-12-18 15:56:51 +00:00
Wim Taymans
2ea251a366 gst-libs/gst/audio/gstbaseaudiosrc.c: Add debug info.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Add debug info.
When going from PLAYING to PAUSED, pause the ringbuffer before calling
the parent state change function, just like the audiosink, because the
parent waits for the element to finish its processing before completing
the state change. This makes going to PAUSED a lot snappier.
When going from READY to PAUSED, don't allow the ringbuffer to start
yet.
2007-12-17 16:44:51 +00:00
Wim Taymans
ac1cc82165 gst-libs/gst/audio/gstbaseaudiosink.c: Our EOS time contains the base_time, _wait_eos() expects a running_time so we ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain):
Our EOS time contains the base_time, _wait_eos() expects a running_time
so we have to subtract the base_time again before calling the function.
This fixes an EOS regression where the base_time was added twice and EOS
took longer and longer in certain situations.
Fixes #498767.
2007-11-21 18:02:21 +00:00
Wim Taymans
157a65b15e Expose methods for some object properties so that subclasses can more easily configure them.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
(gst_base_audio_sink_set_provide_clock),
(gst_base_audio_sink_get_provide_clock),
(gst_base_audio_sink_set_slave_method),
(gst_base_audio_sink_get_slave_method),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_drain),
(gst_base_audio_sink_none_slaving),
(gst_base_audio_sink_handle_slaving):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Expose methods for some object properties so that subclasses can more
easily configure them.
Added slave method none, that completely disables slaving to the
internal clock.
API: gst_base_audio_sink_set_provide_clock()
API: gst_base_audio_sink_get_provide_clock()
API: gst_base_audio_sink_set_slave_method()
API: gst_base_audio_sink_get_slave_method()
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_set_provide_clock),
(gst_base_audio_src_get_provide_clock),
(gst_base_audio_src_set_property),
(gst_base_audio_src_get_property), (gst_base_audio_src_create):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
Expose methods for some object properties so that subclasses can more
easily configure them.
API: gst_base_audio_src_set_provide_clock()
API: gst_base_audio_src_get_provide_clock()
2007-11-21 13:04:17 +00:00
Ole André Vadla Ravnås
05a205860d gst-libs/gst/audio/gstringbuffer.c: Return NULL instead of an enum that happens to be 0, fixes warning on MSVC (#4921...
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås  <ole.andre.ravnas@tandberg.com>
* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
(gst_ring_buffer_parse_caps):
Return NULL instead of an enum that happens to be 0, fixes warning
on MSVC (#492114).
* gst-libs/gst/audio/gstringbuffer.h:
No trailing commas in enum list (for gcc-2.9x).
* gst/videotestsrc/videotestsrc.c: (random_char):
Make information loss explicit instead of implicitly truncating to
eight bits via the return value.  Fixes runtime error on MSVC when
using the debug CRT (#492114).
* win32/common/config.h.in:
Fix a bunch of '#undef FOO bar', which MSVC doesn't like (#492114).
* win32/common/libgstinterfaces.def:
* win32/common/libgstrtp.def:
Export a few more symbols (#492114).
2007-11-01 12:51:57 +00:00
Stefan Kost
28b46c1e5d gst-libs/gst/audio/audio.*: Readd the deprecation guards, but preserve compilability.
Original commit message from CVS:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/audio.h:
Readd the deprecation guards, but preserve compilability.
2007-11-01 08:06:13 +00:00
Tim-Philipp Müller
55a3eaafea gst-libs/gst/audio/audio.h: Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or compilation with DISABLE_...
Original commit message from CVS:
* gst-libs/gst/audio/audio.h:
Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or
compilation with DISABLE_DEPRECATED and without REMOVE_DEPRECATED
(ie. normal cvs builds) will fail.
2007-10-31 15:30:15 +00:00
Stefan Kost
e37568c196 tell gtk-doc about the deprecation guard. Apply more doc fixes.
Original commit message from CVS:
* docs/libs/Makefile.am:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/audio.h:
* gst-libs/gst/interfaces/mixer.c:
tell gtk-doc about the deprecation guard. Apply more doc fixes.
2007-10-31 12:47:41 +00:00
Stefan Kost
ffa52e2eac Fix the docs according to what gtk-doc complained about.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/interfaces/mixer.h:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
* gst-libs/gst/sdp/gstsdpmessage.c:
Fix the docs according to what gtk-doc complained about.
2007-10-30 20:32:14 +00:00
Wim Taymans
6a20747e83 gst-libs/gst/audio/gstbaseaudiosink.c: Also explicitly release the ringbuffer when going to NULL because it is requir...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_change_state):
Also explicitly release the ringbuffer when going to NULL because it
is required in the setcaps function, before the state change to PAUSED
completes.
2007-10-16 15:33:31 +00:00
Wim Taymans
02f280a9a0 gst-libs/gst/audio/gstbaseaudiosink.c: Use new basesink method to make our EOS drain interruptable.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain):
Use new basesink method to make our EOS drain interruptable.
2007-10-10 15:36:56 +00:00
Wim Taymans
c3dda05a8b gst-libs/gst/audio/gstbaseaudiosrc.c: Also handle the case where there is no clock set on the audio source, like in t...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Also handle the case where there is no clock set on the audio source,
like in the unit tests.
2007-10-08 18:02:53 +00:00
Wim Taymans
5ba1ed3a21 gst-libs/gst/audio/gstbaseaudiosink.c: When slaved to the clock, don't try to align a sample with the previous one wh...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_async_play):
When slaved to the clock, don't try to align a sample with the previous
one when going to PLAYING again.
2007-10-02 11:11:13 +00:00
Jan Schmidt
d5996e9c37 Fix a bunch of compile warnings shown with Forte.
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
(gst_text_overlay_set_property):
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
(gst_rtcp_unix_to_ntp):
* gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_get_type):
* gst/playback/gstqueue2.c:
* tests/examples/seek/seek.c: (set_scale):
Fix a bunch of compile warnings shown with Forte.
* gst/audiorate/gstaudiorate.c:
Always pull in config.h before including any system headers.
2007-09-17 17:24:55 +00:00
Wim Taymans
4764e6044f gst-libs/gst/audio/gstbaseaudiosink.c: Disable pull mode scheduling, we're not ready for it yet and it subtly breaks ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_init):
Disable pull mode scheduling, we're not ready for it yet and it subtly
breaks a lot of things.
2007-09-13 22:52:09 +00:00
Wim Taymans
c942252430 gst-libs/gst/audio/gstbaseaudiosrc.*: Allow othe clocks than the internal clock to be used for the pipeline.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init), (gst_base_audio_src_init),
(gst_base_audio_src_provide_clock),
(gst_base_audio_src_set_property),
(gst_base_audio_src_get_property), (gst_base_audio_src_create):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
Allow othe clocks than the internal clock to be used for the pipeline.
Add property to disable clock provide.
API: GstBaseAudioSrc::provide-clock
2007-09-10 22:10:54 +00:00
Wim Taymans
c2460052b3 gst-libs/gst/audio/gstbaseaudiosink.c: When skew slaving, try to hover around the middle of a segment so that we at m...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
When skew slaving, try to hover around the middle of a segment so that
we at most drift by half a segment.
If we are aligning in the oposite direction of the clock skew, we don't
have to resync.
2007-09-03 19:17:33 +00:00
Stefan Kost
a5e777fac3 Original commit message from CVS:
reviewed by: <delete if not using a buddy>
patch by: <delete if not someone else's patch>
* configure.ac:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/gnomevfs/gstgnomevfssrc.h:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst/typefind/gsttypefindfunctions.c:
* gst/volume/gstvolume.c:
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
* pkgconfig/gstreamer-plugins-base.pc.in:
* sys/v4l/v4lsrc_calls.c:
* tests/examples/Makefile.am:
* win32/common/config.h:
2007-08-23 08:33:43 +00:00
Wim Taymans
478a6592de gst-libs/gst/audio/audio.c: Clarify the docs a little.
Original commit message from CVS:
* gst-libs/gst/audio/audio.c:
Clarify the docs a little.
2007-08-22 15:29:04 +00:00
Sebastian Dröge
846ddaa550 gst-libs/gst/audio/audio.h: Use gst_util_uint64_scale() instead of doing the math with double for GST_FRAMES_TO_CLOCK...
Original commit message from CVS:
* gst-libs/gst/audio/audio.h:
Use gst_util_uint64_scale() instead of doing the math
with double for GST_FRAMES_TO_CLOCK_TIME() and
GST_CLOCK_TIME_TO_FRAMES(). For large timestamps this
prevents rounding errors. Fixes #467667.
2007-08-17 15:24:43 +00:00
Jan Schmidt
d5dc054ea3 gst-libs/gst/audio/audio.c: When clipping a buffer with no timestamp, assume it is within the segment without warnings.
Original commit message from CVS:
* gst-libs/gst/audio/audio.c:
When clipping a buffer with no timestamp, assume it is
within the segment without warnings.
Fixes: #460978
2007-07-27 17:10:47 +00:00
Sebastian Dröge
6be2524031 API: Add buffer clipping function for raw audio buffers. Fixes #456656.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/audio.c: (gst_audio_buffer_clip):
* gst-libs/gst/audio/audio.h:
* tests/check/libs/audio.c: (GST_START_TEST), (audio_suite):
API: Add buffer clipping function for raw audio buffers. Fixes #456656.
Also add deprecation guards for gst_audio_structure_set_int() to the
header.
2007-07-23 18:26:09 +00:00
Tim-Philipp Müller
8a499651b9 gst-libs/gst/audio/gstbaseaudiosink.c: Quick hack to make audiosinks stop at EOS when operating in pull-mode; needs t...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_callback):
Quick hack to make audiosinks stop at EOS when operating in
pull-mode; needs to be fixed properly some day.
2007-07-08 13:07:38 +00:00
Andy Wingo
ae6fd1b3f2 gst-libs/gst/audio/gstbaseaudiosink.c
Original commit message from CVS:
2007-06-19  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_init): Enable pull-mode operation.
2007-06-19 19:13:04 +00:00
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
Jan Schmidt
45e06fe704 gst-libs/gst/audio/: Document better the fact that latency_time and buffer_time are values stored in microseconds, an...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_setcaps):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps),
(gst_ring_buffer_samples_done):
* gst-libs/gst/audio/gstringbuffer.h:
Document better the fact that latency_time and buffer_time are values
stored in microseconds, and not the usual GStreamer nanoseconds.
Change the variables (compatibly) that store them from GstClockTime
to guint64 to make it more clear that they're not storing clock times.
Also, remove the bogus property description that says the user can
specify -1 to get the default value, since that's never been the case.
When computing the default segment size for the ring buffer, make it
an integer number of samples.
When the sub-class indicates a delay greater than the number of
samples we've written return 0 from the audio sink get_time method.
2006-06-03 21:06:49 +00:00
Stefan Kost
131fb86b4b Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
Original commit message from CVS:
* ext/alsa/gstalsamixerelement.h:
* ext/alsa/gstalsamixeroptions.h:
* ext/alsa/gstalsamixertrack.h:
* ext/gnomevfs/gstgnomevfssink.h:
* ext/gnomevfs/gstgnomevfssrc.h:
* ext/theora/gsttheoradec.h:
* ext/theora/gsttheoraenc.h:
* ext/theora/gsttheoraparse.h:
* ext/vorbis/vorbisparse.h:
* gst-libs/gst/audio/gstaudioclock.h:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
* gst/audioconvert/gstaudioconvert.h:
* gst/audioresample/gstaudioresample.h:
* gst/audiotestsrc/gstaudiotestsrc.h:
* gst/ffmpegcolorspace/gstffmpegcolorspace.h:
* gst/playback/gststreamselector.h:
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcpclientsink.h:
* gst/tcp/gsttcpclientsrc.h:
* gst/tcp/gsttcpserversink.h:
* gst/tcp/gsttcpserversrc.h:
* gst/videorate/gstvideorate.h:
* gst/videoscale/gstvideoscale.h:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/volume/gstvolume.h:
* sys/v4l/gstv4ljpegsrc.h:
* sys/v4l/gstv4lmjpegsink.h:
* sys/v4l/gstv4lmjpegsrc.h:
* sys/v4l/gstv4lsrc.h:
* sys/ximage/ximagesink.h:
* sys/xvimage/xvimagesink.h:
* tests/old/testsuite/alsa/sinesrc.h:
Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
2006-06-01 19:19:51 +00:00
Tim-Philipp Müller
10d35563dd gst-libs/gst/audio/multichannel.c: It's okay to have caps with channels=1 and a channel position different from GST_A...
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions):
It's okay to have caps with channels=1 and a channel position
different from GST_AUDIO_CHANNEL_POSITION_FRONT_MONO
(deinterleavers might want to keep the position in the caps,
so that they can be re-interleaved again properly later).
Leave check for unexpected 2-channel layouts intact for now.
2006-05-16 17:34:14 +00:00
Stefan Kost
e972defd3e make GstElementDetails const
Original commit message from CVS:
* ext/alsa/gstalsamixerelement.c:
* ext/alsa/gstalsasrc.c:
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/gnomevfs/gstgnomevfssink.c:
* ext/gnomevfs/gstgnomevfssrc.c:
* ext/ogg/gstoggdemux.c:
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c:
* ext/ogg/gstogmparse.c:
* ext/pango/gstclockoverlay.c:
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
* ext/pango/gsttimeoverlay.c:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* gst-libs/gst/audio/gstaudiofilter.c:
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
* gst/audioconvert/gstaudioconvert.c:
* gst/audiorate/gstaudiorate.c:
* gst/audioresample/gstaudioresample.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/playback/gstdecodebin.c:
* gst/playback/gstplaybin.c:
* gst/playback/gststreamselector.c:
* gst/subparse/gstsubparse.c:
* gst/tcp/gstmultifdsink.c:
* gst/tcp/gsttcpclientsink.c:
* gst/tcp/gsttcpclientsrc.c:
* gst/tcp/gsttcpserversink.c:
* gst/tcp/gsttcpserversrc.c:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
* gst/videorate/gstvideorate.c:
* gst/videoscale/gstvideoscale.c:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/volume/gstvolume.c:
* sys/v4l/gstv4ljpegsrc.c:
* sys/v4l/gstv4lmjpegsink.c:
* sys/v4l/gstv4lmjpegsrc.c:
* sys/v4l/gstv4lsrc.c:
* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
* tests/check/libs/cddabasesrc.c:
make GstElementDetails const
2006-04-28 19:46:37 +00:00
Wim Taymans
102b79e46e gst-libs/gst/audio/gstbaseaudiosink.c: patch to make timestamp checking more tollerant to rounding errors given that ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_event),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
patch to make timestamp checking more tollerant to rounding
errors given that real discontinuities are to be marked on
buffers. Fixes some asf files and #338778.
Also avoid some crashers when we receive an event in the
NULL state.
2006-04-28 15:08:09 +00:00
Wim Taymans
04754176a6 gst-libs/gst/audio/gstringbuffer.c: Check arguments passed to public functions instead of crashing.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_set_callback), (gst_ring_buffer_acquire),
(gst_ring_buffer_release), (gst_ring_buffer_is_acquired),
(gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
(gst_ring_buffer_pause), (gst_ring_buffer_stop),
(gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
(gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
(gst_ring_buffer_commit), (gst_ring_buffer_read),
(gst_ring_buffer_prepare_read), (gst_ring_buffer_advance),
(gst_ring_buffer_clear), (gst_ring_buffer_may_start):
Check arguments passed to public functions instead of
crashing.
2006-04-28 14:48:11 +00:00
Wim Taymans
c068425b38 gst-libs/gst/audio/gstbaseaudiosrc.c: GstBaseAudioSrc must be live or it does not work.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
(gst_base_audio_src_get_time), (gst_base_audio_src_create):
GstBaseAudioSrc must be live or it does not work.
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audio_src_init):
Don't set live to TRUE as this is the default in the parentclass.
2006-04-28 14:37:46 +00:00
Wim Taymans
35058f78c1 gst-libs/gst/audio/gstbaseaudiosink.c: Starting the ringbuffer when we did not acquire it can cause a deadlock, is po...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event):
Starting the ringbuffer when we did not acquire it can cause
a deadlock, is pointless and causes nasty things for
subclasses.
Fixes gst-launch audiotestsrc num-buffers=0 ! alsasink.
2006-04-10 17:05:46 +00:00
Stefan Kost
0afac375b4 Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
Original commit message from CVS:
* ext/alsa/gstalsamixeroptions.c:
(gst_alsa_mixer_options_class_init):
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_class_init):
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
* ext/ogg/gstoggparse.c: (gst_ogg_parse_class_init):
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_class_init):
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_class_init):
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_class_init):
* gst-libs/gst/audio/gstaudiosrc.c:
(gst_audioringbuffer_class_init):
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_class_init):
* gst-libs/gst/interfaces/colorbalancechannel.c:
(gst_color_balance_channel_class_init):
* gst-libs/gst/interfaces/mixeroptions.c:
(gst_mixer_options_class_init):
* gst-libs/gst/interfaces/mixertrack.c:
(gst_mixer_track_class_init):
* gst-libs/gst/interfaces/tunerchannel.c:
(gst_tuner_channel_class_init):
* gst-libs/gst/interfaces/tunernorm.c: (gst_tuner_norm_class_init):
* gst-libs/gst/netbuffer/gstnetbuffer.c:
(gst_netbuffer_class_init):
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init):
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init):
* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
* gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
* gst/playback/gststreamselector.c:
(gst_stream_selector_class_init):
* gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
* sys/v4l/gstv4lcolorbalance.c:
(gst_v4l_color_balance_channel_class_init):
* sys/v4l/gstv4ljpegsrc.c: (gst_v4ljpegsrc_class_init):
* sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
* sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
* sys/v4l/gstv4ltuner.c: (gst_v4l_tuner_channel_class_init),
(gst_v4l_tuner_norm_class_init):
* sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
* tests/old/testsuite/alsa/sinesrc.c: (sinesrc_class_init):
Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
2006-04-08 21:02:53 +00:00
Stefan Kost
1a2642a1d2 Fix broken GObject macros
Original commit message from CVS:
* ext/pango/gsttextrender.h:
* gst-libs/gst/audio/gstaudiosink.h:
* gst-libs/gst/audio/gstaudiosrc.h:
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstringbuffer.h:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
* gst-libs/gst/rtp/gstbasertppayload.h:
* gst-libs/gst/video/gstvideofilter.h:
* gst-libs/gst/video/gstvideosink.h:
* gst/playback/gstplaybasebin.h:
* gst/tcp/gstmultifdsink.h:
* sys/v4l/gstv4lelement.h:
Fix broken GObject macros
2006-04-08 18:09:17 +00:00
Stefan Kost
2d826700fa Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init),
(gst_gnome_vfs_sink_class_init):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init):
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init):
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init):
* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
(gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init):
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_base_init):
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_base_init):
* gst/adder/gstadder.c: (gst_adder_get_type):
* gst/adder/gstadder.h:
* gst/audioconvert/gstaudioconvert.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
(gst_audio_test_src_create):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/playback/gstdecodebin.c:
* gst/playback/gstplaybin.c:
* gst/playback/gststreamselector.c:
(gst_stream_selector_base_init):
* gst/subparse/gstsubparse.c: (gst_sub_parse_base_init):
* gst/volume/gstvolume.c:
* sys/v4l/gstv4lmjpegsink.c:
* sys/v4l/gstv4lmjpegsrc.c:
* tests/check/libs/cddabasesrc.c:
* tests/old/examples/gob/gst-identity2.gob:
Add docs for adder, use GST_ELEMENT_DETAILS macro,
define GstElementDetails at the top
2006-03-24 10:42:11 +00:00
Wim Taymans
4df07064b8 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix audio sources, forgot to make the ringbuffer startable...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_get_time), (gst_base_audio_src_create),
(gst_base_audio_src_change_state):
Fix audio sources, forgot to make the ringbuffer
startable...
2006-03-23 16:58:03 +00:00
Wim Taymans
2df1088b3f gst-libs/gst/audio/gstbaseaudiosrc.c: unparent instead of unref the ringbuffer.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_get_time), (gst_base_audio_src_create),
(gst_base_audio_src_change_state):
unparent instead of unref the ringbuffer.
2006-03-23 16:29:58 +00:00
Wim Taymans
227474e464 gst-libs/gst/audio/gstbaseaudiosink.c: Implement new async_play vmethod to start slaving and allow playback start in ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_async_play),
(gst_base_audio_sink_do_play), (gst_base_audio_sink_change_state):
Implement new async_play vmethod to start slaving and allow
playback start in case of async PLAY state changes.
* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
Enable QoS with new method in base class.
2006-03-23 16:24:23 +00:00
Wim Taymans
747d560fb5 gst-libs/gst/audio/gstbaseaudiosink.c: Since we _parent the ringbuffer, we also need to _unparent instead of a plain ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_dispose):
Since we _parent the ringbuffer, we also need to
_unparent instead of a plain _unref.
2006-03-22 12:33:09 +00:00
Wim Taymans
82fd38fbcf gst-libs/gst/audio/: Only start playback if we are playing. should fix #330748.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstringbuffer.c: (wait_segment),
(gst_ring_buffer_may_start):
* gst-libs/gst/audio/gstringbuffer.h:
Only start playback if we are playing.
should fix #330748.
2006-03-17 17:48:33 +00:00
Tim-Philipp Müller
ab6f99ab60 gst-libs/gst/audio/gstaudiofilter.c: Don't ignore flow return from gst_pad_push().
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_chain):
Don't ignore flow return from gst_pad_push().
2006-03-07 13:01:21 +00:00
Christophe Fergeau
8e6d3a5c03 Don't leak references returned by gst_pad_get_parent()
Original commit message from CVS:
* ext/libvisual/visual.c: (gst_visual_getcaps),
(gst_visual_src_setcaps), (gst_visual_sink_setcaps):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_sinkconnect):
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
(gst_vorbisenc_convert_sink):
* gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
(gst_audio_duration_from_pad_buffer):
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_link),
(gst_audio_filter_chain):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps):
* gst-libs/gst/video/video.c: (gst_video_frame_rate),
(gst_video_get_size):
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
Don't leak references returned by gst_pad_get_parent()
(#333663, based on patch by: Christophe Fergeau).
2006-03-07 12:49:03 +00:00
Wim Taymans
1e9f5c43ad docs/: Added some more docs to libs and plugins.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
Added some more docs to libs and plugins.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_prepare_read), (gst_ring_buffer_clear):
* gst-libs/gst/audio/gstringbuffer.h:
Document ringbuffer some more.
* gst/videorate/gstvideorate.c: (gst_video_rate_class_init),
(gst_video_rate_setcaps), (gst_video_rate_reset),
(gst_video_rate_init), (gst_video_rate_flush_prev),
(gst_video_rate_swap_prev), (gst_video_rate_event),
(gst_video_rate_chain), (gst_video_rate_change_state):
* gst/videorate/gstvideorate.h:
Fix videorate to use segments.
Make it work with 0/1 framerates (closes #331903)
Handle EOS correctly.
Added docs.
2006-03-02 16:47:34 +00:00
Wim Taymans
77ff8c9fdb gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock in the NULL state.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_provide_clock):
Don't try to provide a clock in the NULL state.
2006-02-28 11:06:24 +00:00
Tim-Philipp Müller
043c6d91df gst-libs/gst/audio/mixerutils.c: Make order in which elements are tried more determinable.
Original commit message from CVS:
* gst-libs/gst/audio/mixerutils.c:
(element_factory_rank_compare_func):
Make order in which elements are tried more determinable.
2006-02-20 16:21:14 +00:00
Wim Taymans
3451a81879 gst-libs/gst/audio/gstringbuffer.c: Small cleanups.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_get_type),
(gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
(gst_ring_buffer_device_is_open), (gst_ring_buffer_acquire),
(gst_ring_buffer_release), (gst_ring_buffer_set_flushing),
(gst_ring_buffer_start), (gst_ring_buffer_pause_unlocked),
(gst_ring_buffer_pause), (gst_ring_buffer_stop),
(gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
(gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
(gst_ring_buffer_commit), (gst_ring_buffer_prepare_read),
(gst_ring_buffer_clear):
Small cleanups.
Added some G_LIKELY.
2006-02-17 14:07:01 +00:00
Wim Taymans
454618e9b9 gst-libs/gst/audio/TODO: Update TODO
Original commit message from CVS:
* gst-libs/gst/audio/TODO:
Update TODO

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

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

* ext/alsa/gstalsasrc.c (gst_alsasrc_delay): Clamp the delay to
the buffer size -- we don't care about underrun/overrun reporting
right now, just need to return a useful value.
2006-02-09 11:36:18 +00:00
Wim Taymans
260b5295c9 gst-libs/gst/audio/gstbaseaudiosink.c: Ugh.. getting late I guess...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_provide_clock):
Ugh.. getting late I guess...
2006-02-02 18:18:31 +00:00
Wim Taymans
c78a5d7e1e gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock when we are not negotiated since we might not be ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_provide_clock),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_render):
Don't try to provide a clock when we are not negotiated since
we might not be able to make it run.
2006-02-02 18:13:26 +00:00
Wim Taymans
416c011f11 gst-libs/gst/audio/TODO: Updated.
Original commit message from CVS:
* gst-libs/gst/audio/TODO:
Updated.

* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_event):
On EOS, wait till the last sample is played before posting EOS.
2006-02-02 12:14:35 +00:00
Wim Taymans
a169abc679 gst-libs/gst/audio/gstaudiosink.c: Implement pause that does not wait for completion.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_class_init), (gst_audioringbuffer_release),
(gst_audioringbuffer_pause):
Implement pause that does not wait for completion.

* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Don't drop buffers when going to PAUSED but perform preroll on
remaining samples now that core base class supports this.

* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_release),
(gst_ring_buffer_pause_unlocked), (gst_ring_buffer_stop),
(gst_ring_buffer_commit):
Pause should not signal waiters.
Implement return value of _commit correctly.
2006-01-30 16:19:33 +00:00
Sébastien Moutte
dc46970cdf gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
Original commit message from CVS:
* gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
* gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render)
use of gst_guint64_to_gdouble to be compliant with vs6
* gst/playback/gstdecodebin.c: (try_to_link_1)
* gst/videorate/videorate.c: (gst_video_rate_blank_data)
use of G_GINT64_CONSTANT for int64 constants
* win32/common/libgstinterfaces.def:
export some symbols (gst_mixer_get_type,gst_mixer_track_get_type)
* win32/vs6:
update and add new project files
2006-01-29 19:13:39 +00:00
Tim-Philipp Müller
27ed152e10 gst-libs/gst/audio/gstringbuffer.c: Make gcc-4.1 happy (part of #327357).
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format):
Make gcc-4.1 happy (part of #327357).
2006-01-28 18:19:18 +00:00
Wim Taymans
ccd05fa086 gst-libs/gst/audio/gstbaseaudiosink.c: Undo previous commit, it breaks resume after pause.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Undo previous commit, it breaks resume after pause.
2006-01-25 10:50:32 +00:00
Wim Taymans
2bc5ca1786 gst-libs/gst/audio/gstbaseaudiosink.c: Improve debugging.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_setcaps), (gst_base_audio_sink_event),
(gst_base_audio_sink_preroll), (gst_base_audio_sink_render):
Improve debugging.
Post error when caps cannot be parsed.
Resync on discontinuity in the stream.
Clip samples to segment boundaries.
return WRONG_STATE sooner when we are flushing.

* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
(gst_base_audio_src_get_time), (gst_base_audio_src_create):
Make audiosrc operate in TIME.
Set TIMESTAMP and DURATION on buffers.
2006-01-25 09:27:01 +00:00
Jan Schmidt
04333a568c gst-libs/gst/audio/gstbaseaudiosink.c: Fix playback of non-synchronised streams by assuming a rate of 1.0 instead of ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Fix playback of non-synchronised streams by assuming a rate
of 1.0 instead of a random one.

Makes this work again:

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

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

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

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

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

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

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

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

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