The only way for ALSA to expose a position-less multi channels is to
return an array full of SND_CHMAP_MONO. Converting this to a
GST_AUDIO_CHANNEL_POSITION_MONO array would be invalid as
GST_AUDIO_CHANNEL_POSITION_MONO is meant to be used only with one
channel.
Fix this by using GST_AUDIO_CHANNEL_POSITION_NONE which is meant to be
used for position-less channels.
https://bugzilla.gnome.org/show_bug.cgi?id=763799
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.
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
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
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.
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
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
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
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().
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
