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1096 commits

Author SHA1 Message Date
Olivier Crête
b35bc51ed6 audio: Fix annotations 2012-09-13 17:11:56 -04:00
Wim Taymans
0ce33461c8 audiosrc: check for flushing state in provide_clock 
Only provide a clock when we are not flushing, this means that we have posted a
PROVIDE_CLOCK message. We used to check if we were acquired but that doesn't
work anymore now that we do the negotiation async in the streaming thread: it's
possible that we are still negotiating when the pipeline asks us for a clock.
 2012-09-10 12:19:22 +02:00
Wim Taymans
44dab50b7a ringbuffer: add method to check the flushing state 2012-09-10 12:19:22 +02:00
Mark Nauwelaerts
75fe950c33 gst-libs: restore original full padding 2012-09-10 11:45:44 +02:00
Pontus Oldberg
a2f8ec4f5a ringbuffer: add support for timestamps 
Make it possible for subclasses to provide the timestamp (as an absolute time
against the pipeline clock) of the last read data.
Fix up alsa to provide the timestamp received from alsa. Because the alsa
timestamps are in monotonic time, we can only do this when the monotonic clock
has been selected as the pipeline clock.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=635256
 2012-09-10 11:34:14 +02:00
Mark Nauwelaerts
a29fab200c audio{de,en}coder: use GstClockTime parameters where appropriate 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=683672
 2012-09-10 11:20:50 +02:00
Thibault Saunier
dc5bb008a3 audio: port to the new GLib thread API 2012-09-09 20:41:06 -03:00
Tim-Philipp Müller
2079a8c12b Remove glib-compat-private.h stuff we don't need any more 
It's all been ported to the latest GLib API now.
 2012-09-09 18:36:49 +01:00
Mark Nauwelaerts
c9d3f32cc9 audioencoder: plug some leaks 2012-09-06 12:16:59 +02:00
Wim Taymans
668ce33384 update for basesink change 2012-09-04 12:18:11 +02:00
Tim-Philipp Müller
a99a1042b9 gst_message_new_duration() -> gst_message_new_duration_changed() 2012-09-02 01:27:17 +01:00
Jan Schmidt
5dafecad31 audiodecoder: Handle GAP events in place of segment updates 
Use them to trigger generation of an empty output buffer or
to send pending events downstream and trigger pre-roll
 2012-08-31 12:42:12 -07:00
Edward Hervey
def07410ef audiobasesink: Avoid resetting ringbuffer when not needed 
If the ringbuffer was configured to the same caps as previously, we
don't need to reconfigure it.
 2012-08-14 18:56:00 +02:00
Víctor Manuel Jáquez Leal
f7f0c55e5f audiodecoder: getter for allocator 
Sometimes the decoder would use the allocator for something else than just
allocating output buffers, for example, querying different parameters.

This patch expose a getter accessor for the negotiated memory allocator.
 2012-08-14 15:47:34 +02:00
Víctor Manuel Jáquez Leal
936ec3eb8f audioencoder: getter for allocator 
Sometimes the encoder would use the allocator for something else than just
allocating output buffers, for example, querying different parameters.

This patch expose a getter accessor for the negotiated memory allocator.
 2012-08-14 15:47:29 +02:00
Tim-Philipp Müller
2ff4d2efe3 audioencoder: return TRUE from _set_output_format() if all is good 
Fixes not-negotiated errors in wavpackenc unit test.
 2012-08-13 23:34:52 +01:00
Sebastian Dröge
62ec7f837d audioencoder: Let global tag events be handled the same way as other events 2012-08-09 17:06:31 +02:00
Sebastian Dröge
e9fbba63b5 audiodecoder: Let global tag events be handled the same way as other events 2012-08-09 16:55:19 +02:00
Sebastian Dröge
2a1f8a4da3 audio: Merge upstream stream tags 2012-08-09 16:24:47 +02:00
Sebastian Dröge
7f0e65bb46 audio: Always keep a complete taglist around 
Otherwise updates to the tags will cause non-updated
tags to be lost downstream.
 2012-08-09 15:48:03 +02:00
Sebastian Dröge
bc4d923982 audioencoder: Add negotiate vfunc that is used to negotiate with downstream 
The default implementation negotiates a buffer pool and allocator
with downstream.
 2012-08-09 15:27:33 +02:00
Sebastian Dröge
9309272309 audioencoder: Decouple setting of output format and downstream negotiation 
This makes the audio encoder base class more similar to the video
encoder base class.
 2012-08-09 15:21:01 +02:00
Sebastian Dröge
513d4f7cd1 audiodecoder: Add negotiate vfunc that is used to negotiate with downstream 
The default implementation negotiates a buffer pool and allocator
with downstream.
 2012-08-09 15:10:05 +02:00
Sebastian Dröge
e1702d62a0 audiodecoder: Decouple setting of output format and downstream negotiation 
This makes the audio decoder base class more similar to the video
decoder base class.
 2012-08-09 15:02:27 +02:00
Tim-Philipp Müller
6422f2d085 Update .gitignore 2012-08-08 09:06:30 +01:00
Tim-Philipp Müller
ca31913c04 audiocdsrc: update for TOC API change 2012-07-28 11:13:12 +01:00
Sebastian Dröge
99d73c94e9 tag: Update for taglist/tag event API changes 2012-07-28 00:35:02 +02:00
Wim Taymans
683a38ad65 update for new variable names 2012-07-27 15:24:43 +02:00
Wim Taymans
40a0624e99 audio-format: fix shift for 18 bits samples 
The 18bits of the sample are in the LSB so we need to shift them 14 positions to
bring them to 32 bits.
 2012-07-26 15:42:38 +02:00
Mark Nauwelaerts
c91615bd82 audio{de,en}coder: delay input caps processing until processing data 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680614
 2012-07-26 14:35:30 +02:00
Mark Nauwelaerts
28537dc73c audioencoder: avoid setting output caps twice 
... which may not be handled or appreciated well downstream,
e.g. muxers only performing header setup once.
 2012-07-25 15:58:19 +02:00
Mark Nauwelaerts
1f962bc108 audioencoder: also consider filter caps in getcaps 2012-07-25 15:58:19 +02:00
Mark Nauwelaerts
26d74941fb Revert "audioencoder: plug caps ref leak" 
This reverts commit 08ff5899a7.

Was not a leak to begin with as we did not have ownership of caps.
 2012-07-25 12:30:54 +02:00
Mark Nauwelaerts
08ff5899a7 audioencoder: plug caps ref leak 2012-07-25 11:58:26 +02:00
Mark Nauwelaerts
473371f943 audiodecoder: hold caps ref while needed 2012-07-25 11:58:26 +02:00
Mark Nauwelaerts
d55529621c audioencoder: correctly compare audio info positions 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680553
 2012-07-25 11:58:26 +02:00
Mark Nauwelaerts
65ea6dee60 audiodecoder: only arrange to reconfigure if data provided 
... otherwise audio format need not be known already.
 2012-07-24 14:48:59 +02:00
Mark Nauwelaerts
d63a4024b8 audiodecoder: minor doc fix 2012-07-24 12:30:21 +02:00
Wim Taymans
5ff002b47a audio: prefix orc_* functions with audio_orc_* 
To avoid potential conflicts in other modules when statically linking
 2012-07-23 17:16:34 +02:00
Sebastian Dröge
d55d7fdc38 audio: Renegotiate if necessary 
And also correct usage of the base class stream lock.
 2012-07-23 12:01:12 +02:00
Sebastian Dröge
7b06c34868 audiodecoder: Handle allocation query 2012-07-23 11:42:22 +02:00
Sebastian Dröge
0814d38e98 audiodecoder: Add propose_allocation, decide_allocation vfuncs and functions to allocate buffers with information from the allocation query results 2012-07-23 10:28:05 +02:00
Sebastian Dröge
0513d3d9f4 audioencoder: Add propose_allocation, decide_allocation vfuncs and functions to allocate buffers with information from the allocation query results 2012-07-23 10:20:05 +02:00
Edward Hervey
55f692eff6 audiodecoder: Don't assert on pad caps not being set 
The decoder might have been de-activated in the meantime (resulting
in NULL pad caps).

If the decoder really isn't configured, then it will error out further
down when checking whether the GST_AUDIO_INFO_IS_VALID()

https://bugzilla.gnome.org/show_bug.cgi?id=667562
 2012-07-19 10:55:53 +02:00
Evan Nemerson
7a7374f2ef audiometa: add missing array array annotations 2012-07-17 11:07:18 +02:00
Evan Nemerson
17815020fd audio: add missing array and element-type annotations for binary data 2012-07-17 11:06:57 +02:00
Evan Nemerson
fd91104636 audio-channels: add missing array-related annotations 2012-07-17 11:06:47 +02:00
Evan Nemerson
1606028c08 audioencoder: add missing element-type to set_headers method 2012-07-17 11:06:22 +02:00
Edward Hervey
2817bdadc9 libs: Remove "Since" markers and minor doc fixups 2012-07-13 12:11:06 +02:00
Edward Hervey
c9428c96b1 baseaudiosink: Resync when ringbuffer resets 
When the ringbuffer gets restarted (like in setcaps), we *will* have
to resync against the new values.

Without this we end up blindly assuming the new samples align to the
old ones.
 2012-07-12 09:51:35 +02:00
Sebastian Dröge
9de1b170b3 audiocdsrc: Remove the TOC query handling 2012-07-05 12:35:35 +02:00
Sebastian Dröge
0ac1596d8d audiocdsrc: Update for TOC API changes 2012-07-05 12:29:00 +02:00
Sebastian Dröge
b362ec3a57 audiocdsrc: Only push TOC event, the TOC message is handled by the sinks 2012-07-03 17:31:54 +02:00
Tim-Philipp Müller
df70b2d2ce audiocdsrc: send TOC event downstream if we're in continuous mode 
If we're in continuous mode where we'll play the entire CD from
start to finish, send a TOC event downstream so any downstream
muxers can write a TOC to indicate where the various tracks
start and end.
 2012-06-28 23:41:16 +01:00
Tim-Philipp Müller
b27c649a48 audiocdsrc: post TOC message on the bus on start-up 
First attempt at implement the various GstToc API
bits in GstAudioCdSrc.

https://bugzilla.gnome.org/show_bug.cgi?id=668996
 2012-06-26 19:53:35 +01:00
Tim-Philipp Müller
a821d428bb audio: make sure g-i doesn't parse orc-generated gstaudiopack.h file 2012-06-24 00:28:40 +01:00
Wim Taymans
c003efcc63 audiobasesink: fix for basesink API change 2012-06-18 11:40:36 +02:00
Jan Schmidt
d9740bf9ba audio decoder: Add some debug output for bad caps from children 2012-06-12 23:52:35 +10:00
Vincent Penquerc'h
f8b8711081 audiodecoder: push queued events only when we have a first buffer 
https://bugzilla.gnome.org/show_bug.cgi?id=675812
 2012-06-11 11:29:13 +01:00
Wim Taymans
9d6967fe9a Add generated orc files 2012-06-08 17:57:43 +02:00
Wim Taymans
12ac9f0aa2 Also build the orc generated code 2012-06-08 17:57:43 +02:00
Wim Taymans
3f8c5ea036 audio: add orc enabled pack and unpack functions 2012-06-08 17:57:43 +02:00
Wim Taymans
8e393d898a audio: add flag to mark possible unpack formats 
Make a new flag to mark formats that can be used in pack and unpack functions.
Mark S32NE and F64NE as those unpack formats
 2012-06-08 17:57:43 +02:00
Sebastian Dröge
462c4cc3d8 audio: Remove unused, generated marshallers 2012-06-08 11:28:56 +02:00
Wim Taymans
3da0b71876 audio: split audio header into logical parts 2012-06-08 10:10:08 +02:00
Wim Taymans
a2172bdb4b update for tag event change 2012-06-06 13:05:47 +02:00
Sebastian Dröge
2667d4bb82 Revert "audiodecoder: Error out earlier in a few places if something goes wrong" 
This reverts commit eb68a2d5a7.

This sometimes errors out too early now, needs some more thoughts.
 2012-06-04 10:01:42 +02:00
Sebastian Dröge
f609b3a627 audiodecoder: Return setcaps return value instead of always TRUE 2012-06-04 09:56:30 +02:00
Sebastian Dröge
eb68a2d5a7 audiodecoder: Error out earlier in a few places if something goes wrong 2012-06-02 17:16:13 +02:00
Wim Taymans
c66da2c74b audio: add flags for the pack/unpack functions 
Add a flag argument to the pack and unpack function so that we can expand it
later when needed. We could for example prefer a High Quality pack/unpack
operation later.
 2012-05-29 09:54:43 +02:00
Arun Raghavan
9c29cd70ee audio: Fix DTS IEC61937 payloading 
DTS type I-III specify the burst length in bits. Only type IV (which we
do not currently support) needs it to be specified in bytes. Thanks to
Julien Moutte for pointing this out.
 2012-05-25 12:38:32 +02:00
Sebastian Rasmussen
b7b123964b gst-libs: make pkg-config get path to pkg-config dirs from configure 
When --with-pkg-config-path is supplied to configure this path is now
explicitly propagated to pkg-config.

https://bugzilla.gnome.org/show_bug.cgi?id=673377
 2012-05-05 23:26:20 +01:00
Sebastian Dröge
69b18ab09d gst-libs: Remove interfaces libs and mixer/tuner interfaces 
The navigation interface is now in the video library.
 2012-04-13 13:14:13 +02:00
Alban Browaeys
6c8abf24cf libs: Link against internal tag library 2012-04-11 09:58:49 +02:00
Sebastian Dröge
8091546694 audio: Remove obsolete FIXME 0.11 2012-04-11 09:57:35 +02:00
Alessandro Decina
ebf80977c4 audiodecoder: don't discard timestamps when consecutive input buffers have the same ts 
Avoid pushing out buffers with the same timestamp only if the out buffers are
decoded from the same input buffer. Instead keep the timestamps when upstream
pushes consecutive buffers with the same ts.
 2012-04-05 10:19:46 +02:00
Mark Nauwelaerts
6eeca397fc audioencoder: plug a definite and rare leak 2012-04-04 19:57:35 +02:00
Sebastian Dröge
65307dd132 gst: Update versioning 2012-04-04 14:55:15 +02:00
Mark Nauwelaerts
91aa1eb7dd audio{de,en}coder: fixup documentation 2012-04-02 14:23:33 +02:00
Sebastian Dröge
b701534204 audioencoder: Fix handling of offset/offset-end for Ogg codecs 
Fixes the vorbisenc unit test.
 2012-03-31 12:55:15 +02:00
Sebastian Dröge
a103fa85a9 audio{en,de}coder: Track input and output segments separately 
They can go out of sync for some time if processing of buffers
on the old segment happens after the segment was received.
 2012-03-30 13:21:09 +02:00
Sebastian Dröge
9cd9f00799 audioencoder: Add gst_audio_encoder_set_headers() to the docs 2012-03-30 12:57:02 +02:00
Sebastian Dröge
78bcb67ea5 audioencoder: Add function to set in-stream headers 
API: gst_audio_encoder_set_headers()

This makes the hack in vorbisenc and probably others in ::pre_push()
unnecessary.
 2012-03-30 12:47:28 +02:00
Sebastian Dröge
f791ec1f10 audioencoder: Rename ::event() to ::sink_event() and add ::src_event() 2012-03-30 12:23:13 +02:00
Sebastian Dröge
d8cb235fe4 audiodecoder: Rename ::event() to ::sink_event() and add ::src_event() 2012-03-30 12:23:13 +02:00
Sebastian Dröge
40a4f2f8aa audiodecoder: Rename _byte_time() to _estimate_rate() 
Which is telling more about what this actually does and is more
consistent with the video base classes.
 2012-03-30 11:51:47 +02:00
Mark Nauwelaerts
2ddc6bb63d audiodecoder: handle downstream seeking query 
... or not, in line with how segment events are treated.
 2012-03-28 16:41:01 +02:00
Wim Taymans
77a4f5865b audioencoder: avoid caps copy 2012-03-27 15:44:43 +02:00
Wim Taymans
32bd12dba9 Merge branch 'master' into 0.11 
Conflicts:
	.gitignore
	common
	configure.ac
	ext/vorbis/gstvorbisdeclib.h
	gst-libs/gst/audio/gstaudioencoder.c
	gst-libs/gst/riff/riff-read.c
	gst/playback/gstplaysink.c
	gst/playback/gstplaysinkconvertbin.c
	tests/check/libs/video.c
 2012-03-22 11:35:13 +01:00
Wim Taymans
a619d3a8b0 update for memory api changes 2012-03-20 13:20:36 +01:00
Mark Nauwelaerts
278b0f093b audio: include audio enumtypes 2012-03-19 16:18:56 +01:00
Wim Taymans
dfb8e7cb2c don't pass random pointers to pull_range 2012-03-16 21:46:47 +01:00
Wim Taymans
4e1ed6f649 audio: fix debug line 2012-03-13 12:39:52 +01:00
Wim Taymans
25137962ad fix for caps API changes 2012-03-11 19:04:41 +01:00
Wim Taymans
7296ef7c63 audiobasesink: add some G_LIKELY 2012-03-09 17:15:38 +01:00
Wim Taymans
94869bff38 audio: avoid buffer copy when nothing is clipped 
when nothing is clipped, return the input buffer instead of creating and
returning an identical copy.
 2012-03-09 16:17:54 +01:00
Sebastian Dröge
7ff608889a audio{en,de}coder: Add optional open/close vfuncs 
This can be used to do something in NULL->READY, like checking
if a hardware codec is actually available and to error out early.
 2012-03-09 10:56:07 +01:00
Tim-Philipp Müller
29c266ccff Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	common
	docs/libs/gst-plugins-base-libs.types
	ext/pango/gsttextoverlay.c
	ext/vorbis/gstvorbisdec.c
	gst/playback/gstplaysink.c
	gst/playback/gstplaysinkconvertbin.c
	sys/ximage/ximagesink.c
	sys/xvimage/xvimagesink.c
 2012-03-08 20:31:34 +00:00
Mark Nauwelaerts
8a3f818dce audiodecoder: add some tag handling convenience help 2012-03-06 16:17:37 +01:00
Mark Nauwelaerts
5a0fff76f3 audiodecoder: add baseclass _CAST macro 2012-03-06 16:17:33 +01:00
Mark Nauwelaerts
d19f5467cc audio: add helper function to convert mask to channel positions 
... as there may be other than raw audio formats using a channel mask,
and there is already one to convert the other way around.
 2012-03-05 13:03:57 +01:00
Mark Nauwelaerts
debbc75272 audioencoder: stop proxying some old-style 0.10 raw audio caps fields 2012-03-05 13:03:57 +01:00
Mark Nauwelaerts
1a2863bf33 audioencoder: store segment event as pending event to forego dropping it 2012-03-05 13:03:57 +01:00
Mark Nauwelaerts
aae64c40a8 audiodecoder: plug caps leak when setting output format 2012-03-05 13:03:57 +01:00
Mark Nauwelaerts
3b0a2a60da audiodecoder: enhance some debug statement 2012-03-05 11:04:20 +01:00
Sebastian Dröge
f7939bb43f Merge branch 'master' into 0.11 
Conflicts:
	NEWS
	RELEASE
	configure.ac
	docs/plugins/gst-plugins-base-plugins.args
	docs/plugins/gst-plugins-base-plugins.hierarchy
	docs/plugins/gst-plugins-base-plugins.interfaces
	docs/plugins/inspect/plugin-adder.xml
	docs/plugins/inspect/plugin-alsa.xml
	docs/plugins/inspect/plugin-app.xml
	docs/plugins/inspect/plugin-audioconvert.xml
	docs/plugins/inspect/plugin-audiorate.xml
	docs/plugins/inspect/plugin-audioresample.xml
	docs/plugins/inspect/plugin-audiotestsrc.xml
	docs/plugins/inspect/plugin-cdparanoia.xml
	docs/plugins/inspect/plugin-encoding.xml
	docs/plugins/inspect/plugin-ffmpegcolorspace.xml
	docs/plugins/inspect/plugin-gdp.xml
	docs/plugins/inspect/plugin-gio.xml
	docs/plugins/inspect/plugin-gnomevfs.xml
	docs/plugins/inspect/plugin-libvisual.xml
	docs/plugins/inspect/plugin-ogg.xml
	docs/plugins/inspect/plugin-pango.xml
	docs/plugins/inspect/plugin-playback.xml
	docs/plugins/inspect/plugin-subparse.xml
	docs/plugins/inspect/plugin-tcp.xml
	docs/plugins/inspect/plugin-theora.xml
	docs/plugins/inspect/plugin-typefindfunctions.xml
	docs/plugins/inspect/plugin-uridecodebin.xml
	docs/plugins/inspect/plugin-videorate.xml
	docs/plugins/inspect/plugin-videoscale.xml
	docs/plugins/inspect/plugin-videotestsrc.xml
	docs/plugins/inspect/plugin-volume.xml
	docs/plugins/inspect/plugin-vorbis.xml
	docs/plugins/inspect/plugin-ximagesink.xml
	docs/plugins/inspect/plugin-xvimagesink.xml
	gst-libs/gst/app/gstappsink.c
	gst-libs/gst/audio/mixer.c
	gst-libs/gst/audio/mixer.h
	gst-libs/gst/tag/gstxmptag.c
	gst-libs/gst/video/colorbalance.c
	gst-libs/gst/video/colorbalance.h
	gst/adder/gstadder.c
	gst/playback/gstplaybasebin.c
	gst/playback/gstplaybin2.c
	gst/playback/gstplaysink.c
	gst/videoscale/gstvideoscale.c
	tests/check/elements/videoscale.c
	tests/examples/seek/seek.c
	tests/examples/v4l/probe.c
	win32/common/_stdint.h
	win32/common/audio-enumtypes.c
	win32/common/config.h
 2012-03-02 10:00:55 +01:00
Wim Taymans
502c12f827 update for metadata API changes 2012-02-29 17:25:10 +01:00
Wim Taymans
a232714065 meta: add return value to transform 2012-02-28 16:18:30 +01:00
Wim Taymans
1c05eeece5 update for metadata tags 2012-02-28 12:10:14 +01:00
Philippe Normand
63ace8872d audio: link against libm 
It is used in gststreamvolume.
 2012-02-27 14:36:25 +00:00
Edward Hervey
59918e841f Suppress deprecation warnings in selected files, for g_value_array_* mostly 2012-02-27 14:28:15 +01:00
Wim Taymans
5a0354b416 audioencoder: don't leak event 2012-02-27 13:08:36 +01:00
Wim Taymans
15eb385412 audioencoder: use default event function 
Implement a default event function so that subclasses can call it without having
to return FALSE (and make it impossible to report errors).
 2012-02-27 12:49:52 +01:00
Wim Taymans
525f330142 update for metadata changes 2012-02-24 10:26:04 +01:00
Wim Taymans
268d52fd33 Merge branch 'master' into 0.11 
Conflicts:
	gst-libs/gst/rtsp/gstrtspconnection.c
	win32/common/libgstaudio.def
 2012-02-17 23:46:17 +01:00
Tim-Philipp Müller
0f6c8a27a7 docs: add new audio base class API to docs and .def file 2012-02-17 15:08:36 +00:00
Wim Taymans
e44dd9db8f Merge branch 'master' into 0.11 
Conflicts:
	gst-libs/gst/audio/gstaudioencoder.c
	gst-libs/gst/pbutils/gstdiscoverer.c
 2012-02-16 14:23:28 +01:00
Mark Nauwelaerts
439884d628 audiodecoder: add some properties to tweak baseclass behaviour 
... so subclass can also rely upon never being bothered with some NULL buffer
it can't do any interesting with, or with any data before it received
any format configuration (and setup properly).
 2012-02-16 12:35:53 +01:00
Mark Nauwelaerts
5b4dc02523 audioencoder: add some properties to tweak baseclass behaviour 
... so subclass can also rely upon never being bothered with less data
than it desires or with some NULL buffer it can't do any interesting with.
 2012-02-16 12:35:51 +01:00
Mark Nauwelaerts
95306e8fef audiodecoder: assert some more that subclass parsed frame has proper len 2012-02-16 12:35:40 +01:00
Wim Taymans
c7d0fb556f audiodecoder: chain up to parent for defaults 
Chain up to the parent instead of using the FALSE return value from
the event function (because it's otherwise impossible to return an error).
 2012-02-15 13:42:19 +01:00
Wim Taymans
b2fbb2e587 audiodecoder: call default event handler 
Call the default event handler for unknown events.
 2012-02-15 13:03:59 +01:00
Wim Taymans
a75e9102c5 GST_FLOW_WRONG_STATE -> GST_FLOW_FLUSHING 2012-02-08 15:17:49 +01:00
Mark Nauwelaerts
97d60612a4 audiodecoder: remove stray obsolete declaration 2012-02-06 22:10:28 +01:00
Mark Nauwelaerts
2bf1a4428e audio: correctly fill in fallback channel positions in stereo case 2012-02-06 22:10:28 +01:00
Wim Taymans
6c08f53416 audiofilter: configure info after calling vmethod 
First call the vmethod and then configure the audioinfo in the baseclass. This
allows subclasses to know about the old format.
 2012-02-06 13:23:26 +01:00
Wim Taymans
fe3e9b90dd audioencoder: don't unref caps parameter 
Fix refcounting on incomming caps to make sure we don't unref it too much.
 2012-02-03 09:51:00 +01:00
Sebastian Dröge
1cb4029d00 audioencoder: gst_pad_get_pad_template_caps() now returns a new reference, don't forget to unref 2012-02-01 16:33:30 +01:00
Sebastian Dröge
5aa6748151 audio{enc,dec}oder: Check if srcpad caps are a subset of the template caps 2012-02-01 16:32:53 +01:00
Sebastian Dröge
0370b0dc12 audioencoder: Add gst_audio_encoder_set_output_format() function for consistency 2012-02-01 16:27:47 +01:00
Sebastian Dröge
dbd43c7dd3 audiodecoder: Rename set_outcaps() to set_output_format() and take a GstAudioInfo as parameter 2012-02-01 16:27:47 +01:00
Wim Taymans
30af2fe7d6 audiosrc: wait on the right cond variable 
This broke with a merge commit
 2012-01-27 18:27:26 +01:00
Wim Taymans
fcdc385aa1 port to new map API 2012-01-25 12:30:53 +01:00
Sebastian Dröge
68c0790817 Merge branch 'master' into 0.11 
Conflicts:
	gst-libs/gst/interfaces/propertyprobe.c
	sys/xvimage/xvimagesink.c
 2012-01-25 11:50:54 +01:00
Wim Taymans
3d42f0f6ed port to new glib thread API 2012-01-19 11:36:17 +01:00
Tim-Philipp Müller
576bbb4fd8 Remove compatibility code cruft for old GLib versions 2012-01-18 17:22:21 +00:00
Mark Nauwelaerts
3e312e6e16 baseaudiosink: commit correct number of samples when not syncing 2012-01-17 21:46:58 +01:00
Mark Nauwelaerts
974c678ec8 audiodecoder: register state change function 2012-01-17 11:53:51 +01:00
Sebastian Dröge
de19cfdd8a audio: More UNPOSITION flag sanity checks 
..and turn the GST_WARNING() into a g_warning(). This is a programming
error and should be fixed.
 2012-01-11 10:49:49 +01:00
Sebastian Dröge
a03f70e3cd audio: Add validity check for the UNPOSITIONED audio flag 
Also reset the flag when parsing caps.
 2012-01-11 10:44:37 +01:00
Sebastian Dröge
05beab5382 audiometa: Improve GstAudioDownmixMeta to be actually usable 
This now has a two-dimensional array of coefficients
as required and also stores the source and destination
channel positions.
 2012-01-10 12:46:05 +01:00
Sebastian Dröge
67c8b0dfbd audio: Don't crash if NULL positions are passed to gst_audio_info_set_format() 2012-01-10 12:02:56 +01:00
Sebastian Dröge
5cb3d75dbf audiobasesink: Fix infinite recursion by chaining up to the correct parent class vfunc 2012-01-09 14:19:54 +01:00
Sebastian Dröge
bb3eb93ee9 audio: Don't check for channel positions in valid order when converting to a channel mask 2012-01-09 08:24:23 +01:00
Edward Hervey
82da418201 audio: Fix size check 
We fail (and return) if the size is *NOT* a multiple of samples.
 2012-01-06 15:14:59 +01:00
Wim Taymans
dd43d0697e audio: expose API to convert channel array to a mask 2012-01-05 13:59:32 +01:00
Sebastian Dröge
9e072ea844 audio: Improve/fix handling of NONE layouts 2012-01-05 10:34:25 +01:00
Sebastian Dröge
8dcea5d498 audio: Add support again for more than 64 channels with NONE layouts 2012-01-05 10:34:25 +01:00
Sebastian Dröge
31c9f7d09a audio: Fix GST_AUDIO_CHANNEL_POSITION_MASK macro 2012-01-05 10:34:25 +01:00
Sebastian Dröge
9d56bf7712 audioencoder: Proxy the channel mask field instead of the old channel-layout field 2012-01-05 10:34:24 +01:00
Sebastian Dröge
8fe5dc53e0 audiocdsrc: Add the layout field to the caps 2012-01-05 10:34:24 +01:00
Sebastian Dröge
810bfec656 audio: Add "layout" field to the raw audio caps 
This can be used to differentiate between interleaved
and non-interleaved audio and whatever comes in the future.
 2012-01-05 10:34:24 +01:00
Sebastian Dröge
e2c6b8ec4d audio: Add function to reorder channel positions from any order to the GStreamer order 2012-01-05 10:34:24 +01:00
Sebastian Dröge
bd40936409 audioringbuffer: Use new function to get a channel reordering map 2012-01-05 10:34:24 +01:00
Sebastian Dröge
9e930a1ade audio: Add documentation for the new functions 2012-01-05 10:34:24 +01:00
Sebastian Dröge
c9c12372a5 audio: Add public functions to check channel positions validity and to get a reorder map 2012-01-05 10:34:24 +01:00
Sebastian Dröge
225238a913 audioringbuffer: Add support for reordering of channels 2012-01-05 10:34:16 +01:00
Sebastian Dröge
c227f5e77e audio: Add new channel positions and simplify channel expression in the caps 
The available channel positions are all channels from SMPTE 2036-2-2008
(in that order) and DTS Coherent Acoustics, which are basically all 28
channels that currently can appear.

The channels are now expressed in the caps as a channel-mask, which
describes which of the channels are present, and an optional
channel-reorder-map, which must only be used after negotiation for
fixated caps.

For negotiation only the channel-mask and the channel count is relevant
and all elements are expected to handle all reorder maps. Elements that
don't can use the new API to reorder an audio buffer from any order to
another order.

This simplifies negotiation a lot while still having as few reorderings
necassary as possible and still allow all kinds of channel layouts.
 2012-01-05 10:27:21 +01:00
Wim Taymans
e9eaf17eae audioencoder: turn assert into a real error 
Post a real error instead of just asserting. Fixes a unit test.
 2012-01-02 15:42:39 +01:00
Tim-Philipp Müller
26e612aeda playback, mixerutils: gst_registry_get_default() -> gst_registry_get() 2012-01-02 14:32:11 +00:00
Wim Taymans
ed6fd4eb2f audio: add flag for unpositioned layout 
Check if thr layout is explicitly unpositioned and set a flag in the
audio info structure.
 2012-01-02 15:01:58 +01:00
Tim-Philipp Müller
c3e6e23b85 audio, rtsp: remove private/protected gtk-doc markup for enums 
This confuses glib-mkenums, and is not really useful anyway.

https://bugzilla.gnome.org/show_bug.cgi?id=666618
 2012-01-02 00:19:57 +00:00
Tim-Philipp Müller
d877ef13f5 docs: make gtk-doc happier 2011-12-30 19:24:09 +00:00
Tim-Philipp Müller
62e5a67376 audiocdsrc: remove some probing-related vfuncs 
GstPropertyProbe was removed, so these aren't actually used
and we probably want something different for the new API.
 2011-12-30 16:26:47 +00:00
Tim-Philipp Müller
6a85353a92 audiocdsrc: update for GstIndex removal 2011-12-30 16:18:39 +00:00
Tim-Philipp Müller
31890ef59b audiocdsrc: make private bits private 2011-12-30 16:12:30 +00:00
Edward Hervey
f562a29284 Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	ext/theora/gsttheoraenc.c
	gst-libs/gst/tag/gstexiftag.c
	gst/adder/gstadder.c
	gst/adder/gstadder.h
	gst/playback/gstdecodebin2.c
	gst/playback/gstsubtitleoverlay.c
	tests/check/libs/tag.c
 2011-12-30 13:21:35 +01:00
Tim-Philipp Müller
3dfdd6be9d audioringbuffer: rename GST_BUFTYPE_* to GST_AUDIO_RING_BUFFER_FORMAT_TYPE_* 
Bit unwieldy, but more appropriate. Could also be moved into
audio.h as GstAudioFormatType.
 2011-12-25 21:38:21 +00:00
Tim-Philipp Müller
80095caa40 audioringbuffer: remove unused GstAudioRingBufferSegState enum and field 2011-12-25 21:23:11 +00:00
Mark Nauwelaerts
e3c78ff661 audioencoder: add a few more debug statements 2011-12-22 16:58:37 +01:00
Mark Nauwelaerts
9bfa65b7d3 audiodecoder: tweak documentation 2011-12-22 16:58:34 +01:00
Wim Taymans
ddc05e0ed1 propertyprobe: remove propertyprobe 
Remove the propertyprobe interface
Improve docs
 2011-12-21 11:58:53 +01:00
Sebastian Dröge
2760dd2068 audiobasesrc: Use guint8 instead of guchar 2011-12-20 14:36:28 +01:00
Sebastian Dröge
338622fe7e audioringbuffer: Use guint8 instead of guchar 2011-12-20 14:36:28 +01:00
Mark Nauwelaerts
c41f3cbef0 audiodecoder: set a non-zero default maximum tolerated errors 
Whereas the previous default 0 was backwards compatible in that it lead
to erroring out immediately upon any error, elements that are really
ported and using the base class error macro can be assumed to intend to
improve behaviour rather than maintaining the old one.  So, make it easy
on those and any future one and tolerate some errors by default, as intended.

Fixes #666579.
 2011-12-20 12:50:18 +01:00
Wim Taymans
7505b7a55c add audio metadata 
Add some audio metadata to describe a downmix matrix.
Add metadata to media type document.
 2011-12-20 12:02:25 +01:00
Vincent Penquerc'h
12be1e6fc5 baseaudiosink: fix late buffer leak 2011-12-13 12:55:45 +00:00
Tim-Philipp Müller
fb6d09055a Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	ext/alsa/gstalsadeviceprobe.c
	ext/alsa/gstalsamixer.c
	ext/pango/gsttextoverlay.c
	ext/pango/gsttextoverlay.h
	gst-libs/gst/audio/gstaudiobasesink.c
	gst-libs/gst/audio/gstaudioringbuffer.c
	gst-libs/gst/audio/gstaudiosrc.c
	gst-libs/gst/video/Makefile.am
	gst-libs/gst/video/video.c
	gst/encoding/gststreamcombiner.c
	gst/encoding/gststreamsplitter.c
	gst/playback/gstplaybasebin.c
	gst/playback/gststreamsynchronizer.c
	gst/playback/gstsubtitleoverlay.c
	gst/playback/gsturidecodebin.c
	sys/xvimage/xvimagesink.c
	tests/examples/Makefile.am
	win32/common/libgstvideo.def

Video overlay composition disabled for now, needs
porting to buffer meta.
 2011-12-08 01:19:03 +00:00
Wim Taymans
f096b8a8d8 ringbuffer: remove old _full version 2011-12-06 15:06:12 +01:00
Wim Taymans
9e97260c9f fix for basesrc changes 2011-12-06 13:59:11 +01:00
Tim-Philipp Müller
5440ae3c18 Suppress deprecation warnings in selected files, for g_static_rec_mutex_* mostly 
GStaticRecMutex is part of our API/ABI, not much we can do here
in 0.10 for most of these.
 2011-12-04 20:50:25 +00:00
Tim-Philipp Müller
0d98aa25b8 Work around deprecated thread API in glib master 
Add private replacements for deprecated functions such as
g_mutex_new(), g_mutex_free(), g_cond_new() etc., mostly
to avoid the deprecation warnings. We'll change these
over to the new API once we depend on glib >= 2.32.

Replace g_thread_create() with g_thread_try_new().
 2011-12-04 17:16:30 +00:00
Wim Taymans
1225aa9a78 update for basesink event handler changes 2011-12-02 22:24:43 +01:00
Tim-Philipp Müller
177525f89f Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	gst-libs/gst/netbuffer/gstnetbuffer.c
	gst/ffmpegcolorspace/avcodec.h
	gst/ffmpegcolorspace/gstffmpegcodecmap.c
	gst/ffmpegcolorspace/imgconvert.c
	gst/ffmpegcolorspace/imgconvert_template.h
	gst/ffmpegcolorspace/mem.c
	gst/playback/README
	gst/playback/gstplaybasebin.c
	gst/playback/gstplaybasebin.h
	gst/playback/gstplaybin.c
	sys/v4l/v4lmjpegsrc_calls.c
	sys/v4l/videodev_mjpeg.h
	tests/check/elements/gnomevfssink.c
 2011-12-02 11:10:17 +00:00
Piotr Fusik
14644457b0 various: typo fixes 
Fix typos in code and docs. Fixes. #658984
 2011-12-02 12:03:27 +01:00
Wim Taymans
59113af604 Use the new GstSample for snapshots 
Make appsink return a GstSample. Remove the pull_buffer_list method because it
is not very useful anymore.
Pass GstSample to the conversion function.
Update playbin2 and examples
 2011-12-01 16:53:11 +01:00
Edward Hervey
e44db979f9 audio: Add audio-marshal.list to dist-ed files 2011-11-30 11:33:41 +01:00
Wim Taymans
47cbb230e9 audio: move audio interfaces 
Move the audio related interfaces to the audio library.
 2011-11-30 07:57:02 +01:00
Tim-Philipp Müller
0c056a04fe Merge commit '4a58223e4c824fedc024af435337a769e8ce593e' into 0.11 2011-11-28 21:20:10 +00:00
Wim Taymans
5b868bd424 Update for indexable change 2011-11-28 18:24:03 +01:00
Wim Taymans
468d1dde89 audio: update for clock provider API change 2011-11-28 17:51:41 +01:00
Mark Nauwelaerts
4a58223e4c audioencoder: elaborate some documentation 2011-11-28 11:37:33 +01:00
Mark Nauwelaerts
9f57d91137 audiodecoder: add some documentation 2011-11-28 11:37:27 +01:00
Mark Nauwelaerts
856a5dd581 audiodecoder: really discard NULL decoded frame altogether 
... including any timestamp, rather than having that one influence base_ts.
 2011-11-28 11:37:23 +01:00
Tim-Philipp Müller
32b14c6ed3 Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	ext/vorbis/gstvorbisenc.c
	gst/playback/gstdecodebin2.c
	gst/playback/gstplaysinkconvertbin.c
	gst/videorate/gstvideorate.c
 2011-11-26 12:12:59 +00:00
Tim-Philipp Müller
a0639dad38 audio: remove unstable API guards from the audio decoder and encoder base classes 2011-11-25 13:11:54 +00:00
Matej Knopp
817f39608c Fix printf format compiler warnings for OSX / 64bit 
https://bugzilla.gnome.org/show_bug.cgi?id=662607
 2011-11-22 01:00:59 +00:00
Wim Taymans
8fc2a21775 update for activation changes 2011-11-21 13:35:34 +01:00
Wim Taymans
d0bd5f04c0 update for new scheduling query 2011-11-18 17:58:58 +01:00
Wim Taymans
1ad4d20607 add parent to activate functions 2011-11-18 13:56:04 +01:00
Wim Taymans
285702a1a6 fix for scheduling mode rename 2011-11-18 12:37:10 +01:00
Wim Taymans
7afdff3575 Merge branch 'master' into 0.11 
Conflicts:
	gst-libs/gst/audio/gstaudiodecoder.c
 2011-11-17 17:07:41 +01:00
Wim Taymans
e302833e65 add parent to pad functions 2011-11-17 12:48:25 +01:00
Mark Nauwelaerts
69c2c46472 audioencoder: invalidate format info when setup negotiation failed 
... which ensures nothing subsequently tries to slip past _chain
and into a possibly improperly setup subclass.
 2011-11-16 19:03:47 +01:00
Vincent Penquerc'h
f17f918b75 audiodecoder: accept dropped buffers before we know the format 
This allows flacdec to not emit audio for headers, while allowing
the base audio decoder to keep its timestamps in sync.
 2011-11-16 16:54:03 +00:00
Wim Taymans
2202511e77 add parent to query function 2011-11-16 17:25:17 +01:00
Wim Taymans
28157e6f21 _query_peer_*() -> _peer_query_*() 2011-11-15 18:04:17 +01:00
Wim Taymans
ab9ffa93f5 change getcaps to query 
Add sink and src event functions in rtpbasepayload
Add query vmethod to rtpbasepayload.
 2011-11-15 18:04:16 +01:00
Vincent Penquerc'h
3e095382a1 audiodecoder: accept dropped buffers before we know the format 
This allows flacdec to not emit audio for headers, while allowing
the base audio decoder to keep its timestamps in sync.
 2011-11-15 13:29:31 +00:00
Robert Swain
a23dff1fbb audio: Remove some unused variables 2011-11-14 12:49:50 +01:00
Mark Nauwelaerts
38615abdd8 audiodecoder: improve reverse playback 
... by doing some more (reverse) timestamp interpolating and
refactoring downstream pushing.

Fixes #661983.
 2011-11-14 12:00:06 +01:00
Tim-Philipp Müller
c76e5804b3 Update for GstURIHandler get_protocols() changes 2011-11-13 23:44:23 +00:00
Tim-Philipp Müller
455f337e3d gio, appsrc, appsink, cdaudiosrc: update for GstURIHandler API changes 2011-11-13 18:22:06 +00:00
Tim-Philipp Müller
4b0dce5148 Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	gst-libs/gst/audio/Makefile.am
	gst-libs/gst/audio/audio.h
	tests/examples/seek/jsseek.c
	tests/examples/seek/seek.c
	tests/icles/test-colorkey.c
 2011-11-13 13:36:29 +00:00
Tim-Philipp Müller
cd21e69913 audio: add GST_AUDIO_INFO_IS_VALID macro and use in audio decoder base class 
API: GST_AUDIO_INFO_IS_VALID
 2011-11-13 13:18:16 +00:00
Tim-Philipp Müller
394b1f8c3c audio: fix order in LIBADD 
Local libs must come first.
 2011-11-12 12:13:05 +00:00
Tim-Philipp Müller
756c9e2948 audio: fix order in LIBADD 
Local libs must come first.
 2011-11-12 11:58:59 +00:00
Tim-Philipp Müller
dfc13ec632 cdda: rename GstCddaBaseSrc to GstAudioCdSrc and move to libgstaudio 
Another mini-lib down, to make space for new mini libs.

Remove bogus copyright line while at it.
 2011-11-12 11:58:58 +00:00
Wim Taymans
c42e257751 audio: fix docs 2011-11-11 19:13:52 +01:00
Wim Taymans
b645287775 audio: fix headers 
Add const to some methods.
Add padding.
Add GType for GstAudioInfo and GstAudioFormatInfo.
Add new/copy/free for GstAudioInfo.
 2011-11-11 17:53:03 +01:00
Wim Taymans
a3416bc11f rename baseaudio* -> audiobase* 2011-11-11 12:00:52 +01:00
Wim Taymans
ee7072fe7e rename GstBaseAudio* ->GstAudioBase* 2011-11-11 11:52:47 +01:00
Wim Taymans
3d0ac3ded2 rename files to match contained objects 2011-11-11 11:33:15 +01:00
Wim Taymans
6511f36fdb audio: GstRingBuffer -> GstAudioRingBuffer 2011-11-11 11:21:41 +01:00
Wim Taymans
b81af23992 audio: rename internal audio ringbuffer 2011-11-11 10:54:39 +01:00
Wim Taymans
ad8f694ec6 remove bogus files 
They got somehow commited in 7012e88090
 2011-11-11 10:39:52 +01:00
Wim Taymans
e338792ab0 update for adapter api changes 2011-11-10 18:32:39 +01:00
Wim Taymans
f8ef57ca48 Merge branch 'master' into 0.11 2011-11-10 17:26:12 +01:00
Vincent Penquerc'h
0d47c615ad baseaudiosink: make unsigned properties unsigned, not signed 2011-11-10 15:55:31 +00:00
Wim Taymans
57eaf388e0 audio: fix base class vmethods 2011-11-10 16:24:12 +01:00
Wim Taymans
ea9bc40bf9 audiosrc: avoid deadlock 2011-11-10 16:05:19 +01:00
Wim Taymans
1f8fe283f6 audioclock: remove _full version 2011-11-10 13:51:23 +01:00
Wim Taymans
d77c8cafee Merge branch 'master' into 0.11 
Conflicts:
	common
	ext/pango/gsttextoverlay.c
	gst-libs/gst/video/video.c
 2011-11-09 12:11:59 +01:00
Wim Taymans
372b9329b9 remove query types 2011-11-09 11:47:54 +01:00
Tim-Philipp Müller
d7fc45f42e docs: fix up some Since: markers 2011-11-07 23:05:44 +00:00
Wim Taymans
7ac25e9b26 Merge branch 'master' into 0.11 
Conflicts:
	common
	configure.ac
	gst-libs/gst/audio/gstbaseaudiosink.c
	gst/playback/gstdecodebin2.c
	gst/playback/gstplaysinkaudioconvert.c
	gst/playback/gstplaysinkaudioconvert.h
	gst/playback/gstplaysinkvideoconvert.c
	gst/playback/gstplaysinkvideoconvert.h
 2011-11-07 12:23:15 +01:00
Felipe Contreras
3df415d4c7 baseaudiosink: make discont-wait configurable 
Now we can configure how much time to wait before deciding that a
discont has happened.

Also, adds getter and setter to allow derived implementations to set
this value upon construction.

Suggestions and several improvements by Havard Graff.

Signed-off-by: Felipe Contreras <felipe.contreras@gmail.com>
 2011-11-07 11:58:46 +01:00
Felipe Contreras
0a111bf26e baseaudiosink: delay the resyncing of timestamp vs ringbuffertime 
A common problem for audio-playback is that the timestamps might not
be completely linear. This is specially common when doing streaming over
a network, where you can have jittery and/or bursty packettransmission,
which again will often be reflected on the buffertimestamps.

Now, the current implementation have a threshold that says how far the
buffertimestamp is allowed o drift from the ideal aligned time in the
ringbuffer. This was an instant reaction, and ment that if one buffer
arrived with a timestamp that would breach the drift-tolerance, a resync
would take place, and the result would be an audible gap for the
listener.

The annoying thing would be that in the case of a "timestamp-outlier",
you would first resync one way, say +100ms, and then, if the next
timestamp was "back on track", you would end up resyncing the other way
(-100ms) So in fact, when you had only one buffer with slightly off
timestamping, you would end up with *two* audible gaps. This is the
problem this patch addresses.

The way to "fix" this problem with the previous implementation, would
have been to increase the "drift-tolerance" to a value that was greater
than the largest timestamp-outlier one would normally expect.  The big
problem with this approach, however, is that it will allow normal
operations with a huge offset timestamp vs running-time, which is
detrimental to lip-sync. If the drift-tolerance is set to 200ms, it
basically means that lip-sync can easily end up being off by that much.

This patch will basically start a timer when the first breach of
drift-tolerance is detected. If any following timestamp for the next n
nanoseconds gets "back on track" within the threshold, it has basically
eliminated the effect of an outlier, and the timer is stopped.  If,
however, all timestamps within this time-limit are breaching the
threshold, we are probably facing a more permanent offset in the
timestamps, and a resync is allowed to happen.

So basically this patch offers something as rare as both higher
accuracy, it terms of allowing smaller drift-tolerances, as well as much
smoother, less glitchy playback!

Commit message and improvments by Havard Graff.

Fixes bug #640859.
 2011-11-07 11:33:32 +01:00
Felipe Contreras
3f1395afae baseaudiosink: rename some variables 2011-11-07 11:18:34 +01:00
Felipe Contreras
fbde258be6 baseaudiosink: use gst_util_uint64_scale_int when appropriate 
It's probably safer this way.
 2011-11-07 11:11:08 +01:00
Felipe Contreras
369cf3f14a baseaudiosink: split drift-tolerance into alignment-threshold 
So that drift-tolerance is used for clock slaving resync, and
alignment-threshold is for timestamp drift.
 2011-11-07 11:10:05 +01:00
Felipe Contreras
58b9818853 baseaudiosink: trivial comment fixes 
Some found by Havard Graff.

Signed-off-by: Felipe Contreras <felipe.contreras@gmail.com>
 2011-11-07 10:57:56 +01:00
Wim Taymans
2f8292b495 ringbuffer: store bpf in the right variable 2011-11-04 13:21:24 +01:00
Wim Taymans
a5fa136c0b update for tag API removal 2011-11-02 12:11:16 +01:00
Wim Taymans
5bdfd6d899 structure: fix for api update 2011-11-02 09:04:27 +01:00
Tim-Philipp Müller
b52c5819fb Update for pad API changes 
GstProbeType, GstProbeReturn and GstActivateMode -> GstPad*
 2011-11-01 00:34:28 +00:00
Tim-Philipp Müller
220ccdf275 audioencoder: save audio info parsed in setcaps in encoder context 
Otherwise we'll just error out when the first buffer gets pushed.
This is a porting artefact, in 0.10 the infos were allocated on the
heap, now we're doing everything with stack-allocated structs.
 2011-10-31 14:22:39 +00:00
Tim-Philipp Müller
5ee51e47a1 ext, gst, gst-libs, tests: update for tag list API changes 2011-10-31 14:22:39 +00:00
René Stadler
7eb0985282 audio: remove old C file generated from template 
Not sure how this one got pulled into a merge. In 0.10, it was moved away to
gst-template a long time ago. gstaudiofilterexample.c got generated from
gstaudiofiltertemplate.c.
 2011-10-31 15:19:54 +01:00
Wim Taymans
95281cc306 Merge branch 'master' into 0.11 2011-10-28 16:24:44 +02:00