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386 commits

Author SHA1 Message Date
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