Commit graph

158 commits

Author SHA1 Message Date
Axel Mårtensson
5d3c948572 alsasink: pause/resume
alsasink can now detect a resume, stop and pause. The sink is now
properly paused using snd_pcm_pause(), and without losing any data
2019-09-27 05:34:57 +00:00
Thibault Saunier
ac5d0f7da6 alsa: Implement a DeviceProvider
Removing gstalsadeviceprobe.[ch] as it was a relique from the 0.10
century.

This doesn't implement device monitoring but only probing, monitoring
should be implemented in its own commit.
2019-06-04 14:07:37 +00:00
Thibault Saunier
bd72ad601f Mark some properties as DOC_SHOW_DEFAULT 2019-05-13 11:34:08 -04:00
Nicolas Dufresne
d64a4b7a69 Revert "alsa: Implement a DeviceProvider"
This reverts commit 69c3c31608.

All devices have the same name, they are duplicated with pulseaudio one
and the provided does not respond to HW being plugged/unplugged. I think
it's not ready for 1.16.
2019-01-18 11:39:02 -05:00
Thibault Saunier
69c3c31608 alsa: Implement a DeviceProvider
Removing gstalsadeviceprobe.[ch] as it was a relique from the 0.10
century.
2019-01-18 10:18:54 -03:00
Ponnam Srinivas
0e8a510eda alsasink: Fix Memory leak in payload not succuss case
https://bugzilla.gnome.org/show_bug.cgi?id=788114
2017-09-26 11:18:09 +03:00
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
Arun Raghavan
af091928f3 alsa: Trivial doc update
alsasink now does more than just raw audio.
2016-01-25 18:31:17 +05:30
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
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
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
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
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
639e2d4346 alsasink: Update internal buffer/period times with the values that were configured on the device 2013-05-29 16:41:06 +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
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
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
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
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
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
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
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
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
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
Wim Taymans
6511f36fdb audio: GstRingBuffer -> GstAudioRingBuffer 2011-11-11 11:21:41 +01:00
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
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