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