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418 commits

Author SHA1 Message Date
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