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118 commits

Author SHA1 Message Date
Branislav Katreniak
4f88125b3d alsasrc: lock calls to snd_pcm_delay() with mutex as in alsasink 
Alsasrc introduced delay_lock in commit 519f85a43e73efb8f3fb2c7be45226e
because alsa-lib is not thread safe for the same handle.
Alsasrc uses the same threading pattern, it should be locked too.

https://bugzilla.gnome.org/show_bug.cgi?id=746015
 2018-01-22 20:24:00 +00:00
Thibault Saunier
dd1e4f5ebb alsasrc: Handle newly added GstStateChange values 
https://bugzilla.gnome.org/show_bug.cgi?id=783798
 2017-06-20 09:19:53 -04:00
Thibault Saunier
099ac9faf2 docs: Convert gtkdoc comments to markdown 
Modernizing the documentation, making it simpler to read an
modify and allowing us to possibly switch to hotdoc in the
future.
 2017-03-10 18:19:17 -03:00
Guillaume Desmottes
d9268a50f1 alsa: factor out alsa_detect_channels_mapping() 
This code was duplicated in alsasrc and alsasink.

https://bugzilla.gnome.org/show_bug.cgi?id=763985
 2016-04-12 14:34:13 -04:00
Vineeth TM
44b70ca3a1 base: use new gst_element_class_add_static_pad_template() 
https://bugzilla.gnome.org/show_bug.cgi?id=763075
 2016-03-24 14:25:41 +02:00
Nicolas Dufresne
e7a59d2e08 Revert "alsasrc: Disable HW timestamp" 
This reverts commit 3642e9a391.
 2015-12-14 13:59:02 -05:00
Nicolas Dufresne
3642e9a391 alsasrc: Disable HW timestamp 
This is a workaround for broken pulse module.
 2015-12-14 13:39:02 -05:00
Sebastian Dröge
4fe12c1b09 alsa: Use 8 bit pointer type for byte-based pointer arithmetic 
Usually these loops only run once, so there's no problem here. But sometimes
they run twice, and by adding the number of bytes to a 16 bit pointer type we
would advance twice as much as we should.

Also use snd_pcm_frames_to_bytes() in alsasrc to calculate
the number of bytes to skip, same as we do in alsasink.

Thanks to Lucio A. Hernandez <lucio.a.hernandez@gmail.com> for reporting.
 2015-10-14 00:33:49 +03:00
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
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
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
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
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
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
Tim-Philipp Müller
5f59b4f7ee Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2012-11-03 23:05:09 +00:00
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
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
Andoni Morales Alastruey
2434f2932b alsasink: check for spdif support only in the current device 2012-05-18 12:01:06 +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
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
Wim Taymans
dde5e5a248 alsa: remove more property probe stuff 2011-12-22 16:37:29 +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
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
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
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
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