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Author SHA1 Message Date
Antonio Ospite
2c7ed42292 midi: add an ALSA MIDI sequencer source 
The alsamidisrc element allows to get input event from ALSA MIDI
sequencer devices, and possibly convert them to sound using some
downstream element like fluiddec.

Fixes #738687
 2015-10-01 21:43:21 +02:00
Carlos Rafael Giani
8f339c0932 alsa: report recoverable device failures to base class 
This gives custom slave methods in the base class a chance to
resynchronize themselves

Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>

https://bugzilla.gnome.org/show_bug.cgi?id=708362
 2015-06-09 21:51:05 +10:00
Tim-Philipp Müller
ec5c93f169 docs: update element example pipelines 
- gst-launch -> gst-launch-1.0
- use autoaudiosink and audiovideosink more often
- review pipeline examples and descriptions
 2015-05-10 11:38:19 +01:00
Sebastian Dröge
69e338d7dd alsa: Constify channel position table 2015-01-21 09:42:35 +01:00
Sebastian Dröge
8e6fb92886 alsa: Fix indention 2015-01-21 09:42:35 +01:00
Thomas Roos
485ad66a11 alsa: Allow to use 8 bit samples with ALSA 
8 bit samples have no (0) as endianness, not the native endianness.

https://bugzilla.gnome.org/show_bug.cgi?id=739446
 2015-01-21 09:42:35 +01:00
Thomas Klausner
a4b94e6c69 alsa: Use EPIPE instead of ESTRPIPE if the latter does not exist 
NetBSD does not have ESTRPIPE.

https://bugzilla.gnome.org/show_bug.cgi?id=740952
 2014-12-01 09:51:12 +01:00
Sebastian Dröge
90eb93c2ef Don't compare booleans for equality to TRUE and FALSE 
TRUE is 1, but every other non-zero value is also considered true. Comparing
for equality with TRUE would only consider 1 but not the others.
 2014-12-01 09:51:12 +01:00
Tim-Philipp Müller
b6d49d2a12 alsasrc: debug message fixes 
In the same vein as 74e9640a.
 2014-11-25 22:01:08 +00:00
Branislav Katreniak
5e8e6276cd alsa: Change the log messages in xrun_recovery() from DEBUG to WARNING 
xrun_recovery() runs when there is an error

https://bugzilla.gnome.org/show_bug.cgi?id=740615
 2014-11-24 15:30:32 +00:00
Vincent Penquerc'h
b444d8ba97 alsasink: make gst-ident happy 2014-06-03 15:17:20 +01:00
Vincent Penquerc'h
3b2d583373 alsasink: fix occasional crash intersecting invalid values 
When a pipeline using alsasink and push mode upstream fails
to preroll, the following state will be the case:
- A loop upstream will be PAUSED, pushing a first buffer
- alsasink will be READY, pending PAUSED, because async

On error, the pipeline will switch to NULL. alsasink is in
READY, so goes to NULL immediately. It zeroes its cached
caps. Meanwhile, the upstream loop can cause a caps query,
conccurent with the state change. This will use those cached
caps. If the zeroing happens between the NULL test and the
dereferencing, GStreamer will critical down in the GstValue
code.

Since it appears that such a gap between states (PAUSED
and pushing upstream, and NULL downstream) is expected, we
need to protect the read/write access to the cached caps.

This fixes the critical.

See https://bugzilla.gnome.org/show_bug.cgi?id=731121
 2014-06-03 15:17:20 +01:00
Vincent Penquerc'h
74e9640a22 alsasink: pass correct error to g_strerror 
The error we get is a negated errno.

While there, fix a couple typos in messages.
 2014-05-19 13:57:41 +01:00
Tim-Philipp Müller
bcb8068e27 docs: remove outdated and pointless 'Last reviewed' lines from docs 
They are very confusing for people, and more often than not
also just not very accurate. Seeing 'last reviewed: 2005' in
your docs is not very confidence-inspiring. Let's just remove
those comments.
 2014-04-26 23:28:57 +01:00
Sebastian Dröge
d3e0381d3d alsa: Make clang happy with our g_strdup_vprintf() wrapper 2014-02-08 17:01:38 +01:00
Takashi Iwai
76d807893c alsa: Add channel map API support 
The initial support for the new ALSA chmap API.
Just translate the current chmap to GstAudioChannelPosition during the
setup.  No function to specify the channel map manually yet, so still
impossible to assign any non-standard positions or to configure in a
different order even if the hardware allows.

https://bugzilla.gnome.org/show_bug.cgi?id=709755
 2013-10-09 19:05:53 +02:00
Sebastian Dröge
4180581ce9 alsasrc: Dump some more debug output about the device configuration 2013-05-29 16:41:14 +02:00
Sebastian Dröge
639e2d4346 alsasink: Update internal buffer/period times with the values that were configured on the device 2013-05-29 16:41:06 +02:00
Alexander Schrab
a049b102da alsasrc: Make using driver timestamps possible 
https://bugzilla.gnome.org/show_bug.cgi?id=699744
 2013-05-20 11:25:17 +02:00
Sebastian Dröge
0bc25f0325 alsa: Dump min/max period time and buffer time in alsasrc too 2013-05-20 11:23:06 +02:00
Sebastian Dröge
948a4a3632 gst: Add better support for static plugins 2013-04-15 15:52:58 +02:00
yanghuolin
67a7b5a993 alsasink: don't use 100% CPU 
The root cause is that alsa-lib is not thread safe for the same handle.
There are two threads in the gstreamer accessing alsa-lib not serilized.
The race condition happens when one thread holds the old framebuffer app_ptr
position in the kernel, another thread advances the framebuffer app_ptr.
when the former thread is scheduled to run again, it overwrites the app_ptr
to old value by copying from kernel.Thus,the app_ptr in the upper
alsa-lib(pcm_rate) become one period size more advanced than the lower
alsa-lib(pcm_hw & kernel).

gstreamer uses noblock and poll method to communicate with the alsa-lib.
The app_ptr unsync situation as described above makes the poll return immediately because
it concludes there is enough space for the ring-buffer via the low-level alsa-lib.
The write function returns immediately because it concludes there is not enough
space for the ring-buffer from the upper-level alsa-lib. Then the loop of poll
and write runs again and again until another period size is available for
ring-buffer.This leads to the cpu 100 problem.

delay_lock  is used to avoid the race condition.

Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=690937
 2013-01-24 15:08:31 +01:00
Tim-Philipp Müller
df6031f7c6 alsasrc: return negative value on read error 
Otherwise baseaudiosrc won't go into the error code path.

https://bugzilla.gnome.org/show_bug.cgi?id=690197
 2012-12-17 20:50:33 +00:00
Tim-Philipp Müller
3d5a78e67a alsa: post error message when audio device disappears 
Don't loop forever if an USB audio device gets disconnected
while in use. Post an error message instead. This is not
enough yet though, we still need to make the base class
and/or the ring buffer bail out.

https://bugzilla.gnome.org/show_bug.cgi?id=690197
 2012-12-16 01:00:43 +00:00
Sebastian Dröge
d9b25afe71 ext: Fix some compilation errors caused by circular header includes 2012-12-12 17:22:31 +00:00
Tim-Philipp Müller
5f59b4f7ee Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2012-11-03 23:05:09 +00:00
Sebastian Dröge
3c1041d5eb Revert "gst: Add better support for static plugins" 
This reverts commit d2d79e3bc2,
which was accidentially pushed.
 2012-10-24 13:26:26 +02:00
Sebastian Dröge
d2d79e3bc2 gst: Add better support for static plugins 2012-10-24 12:10:44 +02:00
Tim-Philipp Müller
ccbb233da8 alsasink: fix caps leak in acceptcaps function 
https://bugzilla.gnome.org/show_bug.cgi?id=681192
 2012-10-20 11:38:55 +01:00
Tim-Philipp Müller
1a69ec3fd3 alsa: if no formats in native endianness could be detected, try non-native endianness as well 
This can happen, e.g. when using an USB sound card on
a big-endian device

https://bugzilla.gnome.org/show_bug.cgi?id=680904
 2012-10-18 11:04:06 +01:00
Tim-Philipp Müller
1e329bb4f4 alsa: fix supported format detection 
The format probing code was assuming there'd be one caps
structure for each separate width/depth combination like
we did in 0.10 all over the place: for one, we'd query
unsigned/signed formats together for the same width/height,
and we'd add the entire current structure to the probed
caps when we find a format is supported. Now that we have
all raw formats in a single structure, this is all not going
to work so well any more. We added the entire structure with
all possible formats to the caps if we support just one format.

Fix probing so that we only return the list of actually
supported raw audio formats (with native endianness) from
get_caps().
 2012-10-18 11:03:07 +01:00
Arun Raghavan
9f9718715a audio: Explicitly specify endianness for IEC 61937 payloading 
This is required since some systems (DirectSound and OS X) manage the
final byte order themselves.

https://bugzilla.gnome.org/show_bug.cgi?id=678021
 2012-09-19 09:15:16 +05:30
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
Tim-Philipp Müller
794af4fc51 alsa: port to new GLib thread API 2012-09-10 01:06:51 +01:00
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
Tim-Philipp Müller
fc37cf5779 Silence some 'variable may be used uninitialized' compiler warnings 
when compiling with -DG_DISABLE_ASSERT
 2012-08-08 10:19:20 +01:00
Andoni Morales Alastruey
2434f2932b alsasink: check for spdif support only in the current device 2012-05-18 12:01:06 +02:00
Mark Nauwelaerts
1c70c5b85e alsasink: really use local ringbuffer spec helper var and init it a bit more 
... to avoid assertion failures

Conflicts:

	ext/alsa/gstalsasink.c
 2012-05-09 10:28:16 +02:00
Andoni Morales Alastruey
c6409806c1 alsasink: use the iec958 payloader to support non-payloaded input streams 2012-05-07 13:31:01 +02:00
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
Tim-Philipp Müller
3c6a3ad629 Use new gst_element_class_set_static_metadata() 2012-04-10 00:45:16 +01:00
Sebastian Dröge
ad42b16375 gst: Update for GST_PLUGIN_DEFINE() API change 2012-04-05 15:11:05 +02:00
Sebastian Dröge
65307dd132 gst: Update versioning 2012-04-04 14:55:15 +02:00
Wim Taymans
6e054dfc3d alsa: fix small caps leak 2012-03-27 15:43:44 +02:00
Wim Taymans
25137962ad fix for caps API changes 2012-03-11 19:04:41 +01:00
Sebastian Dröge
1cbcb9281c mixer/colorbalance: Update for API changes 2012-03-02 10:00:59 +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
Edward Hervey
59918e841f Suppress deprecation warnings in selected files, for g_value_array_* mostly 2012-02-27 14:28:15 +01:00
Wim Taymans
61a53092e4 alsa: merge instead of appending structures 2012-01-26 14:28:06 +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
Tim-Philipp Müller
5487cb98ef Replace deprecated GStaticMutex with GMutex 2012-01-22 22:52:28 +00:00
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
Vincent Penquerc'h
8d29fe8834 alsasink: fix high sample rates being rejected 
An ALSA sink may select a different rate (as we use the _set_rate_near
API, which is not guaranteed to set the exact target rate).
The rest of the code seems to already handle this well, as output
from a 88200 Hz file seems to have the correct pitch when selecting
a 96 kHz rate.
 2012-01-16 11:46:05 +00:00
Vincent Penquerc'h
361f2b169c alsasink: fix rate match message mistaking error code for sample rate 2012-01-16 11:46:05 +00:00
Vincent Penquerc'h
e60027c795 alsasink: log API errors along with the error code and string 2012-01-16 11:46:05 +00:00
Sebastian Dröge
75f91ebea0 ext: Add new layout field to the raw audio caps 2012-01-05 10:34:25 +01:00
Sebastian Dröge
2fc75efdce alsa: Port to the new multichannel caps 2012-01-05 10:34:20 +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
cab6432c68 alsasink: make work for raw audio formats by fixing template caps 2011-12-23 00:54:43 +00:00
Wim Taymans
dde5e5a248 alsa: remove more property probe stuff 2011-12-22 16:37:29 +01:00
Wim Taymans
ddc05e0ed1 propertyprobe: remove propertyprobe 
Remove the propertyprobe interface
Improve docs
 2011-12-21 11:58:53 +01:00
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
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
4828234639 alsamixer: use GRectMutext instead of GStaticRecMutex with newer glib versions 2011-12-04 20:38:19 +00:00
Tim-Philipp Müller
9c307bccc5 alsamixer: embed static mutexes into the mixer structure 
instead of allocating them dynamically
 2011-12-04 20:21:26 +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
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
Tim-Philipp Müller
ec0d3566bf Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	ext/alsa/gstalsasrc.c
	ext/alsa/gstalsasrc.h
	gst/adder/gstadder.c
	gst/playback/gstplaybin2.c
	gst/playback/gstplaysinkconvertbin.c
	win32/common/libgstvideo.def
 2011-12-02 00:07:39 +00:00
Tim-Philipp Müller
e88e47cd24 Revert "alsasrc: Improve timestamp accuracy" 
This reverts commit 0b774e0b7c.
 2011-11-30 23:15:35 +00:00
Tim-Philipp Müller
e5ae553850 Revert "alsasrc: Fix some compilation errors" 
This reverts commit 2b84f5bd74.
 2011-11-30 23:15:22 +00:00
Tim-Philipp Müller
4cc8920db4 Revert "alsa: Remove unused but set variable" 
This reverts commit e9aed7f31c.
 2011-11-30 23:15:12 +00:00
Tim-Philipp Müller
1290f7de0e Revert "alsasrc: fail gracefully when ALSA does not give timestamps" 
This reverts commit c7282a5718.
 2011-11-30 23:15:03 +00:00
Tim-Philipp Müller
d11849114c Revert "alsasrc: handle the case where the drivers don't supply timestamps" 
This reverts commit 8154b69112.
 2011-11-30 23:14:54 +00:00
Stefan Sauer
6d167abdfa Revert "alsasrc: style fix" 
This reverts commit f70ca6d4cb.
 2011-11-30 23:14:44 +00:00
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
Vincent Penquerc'h
96374054ac various: fix pad template leaks 
https://bugzilla.gnome.org/show_bug.cgi?id=662664
 2011-11-28 13:09:02 +00:00
Stefan Sauer
f70ca6d4cb alsasrc: style fix 
Use timestamp==0 instead of mixing it with !timestamp style checks.
 2011-11-28 10:55:39 +01:00
Stefan Sauer
8154b69112 alsasrc: handle the case where the drivers don't supply timestamps 
If highres-timestamp is 0, try lowres and if that fails fallback to system clock
timestamps.
 2011-11-28 09:13:29 +01:00
Wim Taymans
ee7072fe7e rename GstBaseAudio* ->GstAudioBase* 2011-11-11 11:52:47 +01:00
Wim Taymans
6511f36fdb audio: GstRingBuffer -> GstAudioRingBuffer 2011-11-11 11:21:41 +01:00
Wim Taymans
3254e79f04 alsa: fix negotiation 
Don't assume the format is a string because now it is a list of string in the
template.
Chain up to the parent class implementation of get_caps.
 2011-11-10 16:05:19 +01:00
Wim Taymans
7cd83031a1 alsa: update for new task api 2011-11-02 09:04:27 +01:00
Wim Taymans
06311362e9 fix compilation 2011-10-27 17:26:58 +02:00
Stefan Sauer
53d7d2e966 interfaces: clean up the use of iface and class/klass 2011-10-21 14:46:48 +02:00
Wim Taymans
a00927ad03 Merge branch 'master' into 0.11 2011-10-04 17:58:49 +02:00
Vincent Penquerc'h
c7282a5718 alsasrc: fail gracefully when ALSA does not give timestamps 
https://bugzilla.gnome.org/show_bug.cgi?id=660170
 2011-10-03 11:14:09 +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
Wim Taymans
8023f49d19 more audio caps porting 2011-08-19 17:41:22 +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
Tim-Philipp Müller
c16e7321b9 alsa: don't use GstImplementsInterface 2011-06-26 22:58:17 +01:00
Wim Taymans
2e837743c3 audio: clean up audiosink headers 2011-06-21 18:13:48 +02:00
Wim Taymans
489eed9bb8 Merge branch 'master' into 0.11 2011-05-19 11:31:53 +02:00
Robert Swain
e9aed7f31c alsa: Remove unused but set variable 
Unused but set variables cause warnings in GCC 4.6.x and newer.
 2011-05-18 09:34:52 +02:00
Sebastian Dröge
c255019b90 ext: Update for caps/pad template related API changes 2011-05-17 13:06:01 +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
Sebastian Dröge
0415b90e99 alsa: Update for negotiation related API changes 2011-05-16 15:35:41 +02:00
Sebastian Dröge
2b84f5bd74 alsasrc: Fix some compilation errors 2011-05-14 11:42:32 +02:00