There was no code that included files from other directories using
the #include "..." style before.
Signed-off-by: Alexander E. Patrakov <patrakov@gmail.com>
This makes it easy to log a message every time the reference ratio
changes. I also need to add a hook for reference ratio changes, but
that need will go away if the stream relative volume controls will be
created by the core in the future.
PA_MAYBE_INT16_SWAP() should call PA_INT16_SWAP(), not PA_INT32_SWAP
PA_MAYBE_INT16_SWAP() is not used (yet), so no big deal :)
Signed-off-by: Peter Meerwald <pmeerw@pmeerw.net>
for example, the conversion function for
convert_from_float32ne(PA_SAMPLE_S16LE) can also be used for
convert_to_s16ne(PA_SAMPLE_FLOAT32LE)
v2: ARM can potentially be big- or little endian; only apply
optimization on LE based on WORDS_BIGENDIAN #define (thanks, Tanu)
Signed-off-by: Peter Meerwald <pmeerw@pmeerw.net>
A recent patch broke the build on FreeBSD, which does not have
HAVE_CREDS defined. Also, make sure any attempts to enable the
srbchannel on such architectures fail.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=80642
Reported-by: Ryan Lortie
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
The srbchannel is enabled if protocol version >= 30 and
SHM is available. There is also a module parameter
srbchannel=false that can be used for disabling the srbchannel.
The setup is done in these steps:
1) Server receives authentication (like today)
2) Server sends enable_srbchannel to client
3) Server sends memblock to client
4) Client receives enable_srbchannel
5) Client receives memblock
6) Client sends enable_srbchannel back to server
7) Client switches over
8) Server receives enable_srbchannel and switches over
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
This increments protocol version to v30 and adds two new commands
to enable and disable an shm ringbuffer, as well as client side
implementation.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
For writing, we prefer writing through the srbchannel if one is available,
and we have no ancil data to send.
For reading, we support reading from both in parallel. This meant replicating
a struct used for reading, so a lot of this patch is just a search/replace in
do_read to use the appropriate channel for reading.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
To keep the data and the ringbuffer separate, let's add another
mempool just for the ringbuffer(s). That way, the client can open
the ringbuffer shm file in rw mode and keep the data in ro mode.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
The shared ringbuffer memblock must be writable by both sides.
This makes it possible to send such a memblock over a pstream without
the "both sides writable" information getting lost.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
This is a preparation for the shm ringbuffer, which needs to be able
to be writable by both sides, because there are atomic variables they
both need to modify.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
An shm ringbuffer that is used for low overhead server-client communication.
Signalling is done through eventfd semaphores - it's based on pa_fdsem to avoid
syscalls if nothing is waiting on the other side.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
This patch adds support to iochannel, pstream and pstream-util
to send file descriptors over a unix pipe.
Currently we don't support writing both creds and fds in the same
packet, it's either one or the other (or neither).
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
The file descriptors are read from the iochannel just like the creds are.
So instead of passing just creds (and creds_valid), we now pass the
entire pa_ancil struct.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
To support later patches that add sending/receiving file descriptors,
let's add this struct.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Previously relative cookie paths were searched from the home
directory, now they are searched from the config home directory. This
fixes the problem that XDG_CONFIG_HOME didn't have effect on cookie
paths.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=75006
If a relative path is passed to pa_authkey_load(), it will interpret
the path as relative to the home directory. This is wrong, because
relative paths should be interpreted to be relative to the config home
directory. Before fixing pa_authkey_load(), this patch prepares for
the change by using absolute paths when the file actually needs to be
in the home directory (i.e. the fallback cookie path for the native
protocol and the default cookie path for the esound protocol).
The only place where pa_authkey_load() was called was in
pa_authkey_load_auto(), and the only functionality that
pa_authkey_load() was to log a warning if load() fails. That log
message is now in pa_authkey_load_auto(), so pa_authkey_load() has no
use any more.
This fixes assertion failures that manifest themselves with cards that
support only weird rates such as 37286Hz. Tested with snd-pcsp.
Signed-off-by: Alexander E. Patrakov <patrakov@gmail.com>
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=48109
As it is implemented, the early request mode can in some cases be counter-productive. The mode is designed to give the client a steady request/report rate of small-ish chunks (A somewhat silly client requirement but at least Flash and Firefox break horribly when you break this.).
Unfortunately PulseAudio does not have any mechanism for telling a sink/source how often it should request/report data. So a more blunt hack was applied where the entire latency is restricted to the fragment size.
So far so good, but where the current code breaks down is when the sink cannot satisfy this tiny latency request. We then "report" to the client what we can guarantee by setting the fragment size to the sink's/source's full buffer size/latency.
This severely changes the resulting buffer attributes from what the client requested, and in practice breaks applications. The most prominent user of this feature is the ALSA plugin, and it doesn't even have a mechanism of adapting to the server giving back something different than what was requested.
So long term, the whole early request mode needs to be implemented in a better way. Either the sink's/source's need to grow the ability to control request/report rate. Or we put some form of timer based emulation in front of them on behalf of these clients.
Short term, we should change the behaviour of what happens when we cannot guarantee a fragment rate. Instead of giving the client really shitty buffering parameters as a result, we should just keep the requested attributes and do things on a best-effort basic. Basically how things would behave if the client didn't have the early request bit at all.
The attached patch does just that, as well as expand on the comment about how the early request thing is implemented.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=66962
speex_resample_float() does not work with speex compiled with
--enable-fixed-point, because speex expects its float input
to be normalized to ±32768 instead of the more usual ±1.
It is possible to fix speex_resample_float(), as demonstrated at
http://lists.freedesktop.org/archives/pulseaudio-discuss/2014-May/020617.html
However, a better idea is to avoid using the speex-float resampler and
the associated s16 <-> float conversions that speex will immediately undo
internally if it is known that speex has been compiled with FIXED_POINT.
So, transparently change speex-float-* to speex-fixed-* in that case.
Signed-off-by: Alexander E. Patrakov <patrakov@gmail.com>
Reported-by: Fahad Arslan <fahad_arslan@mentor.com>
Cc: Damir Jelić <poljarinho@gmail.com>
Cc: Peter Meerwald <pmeerw@pmeerw.net>
FIXED_POINT detection is based on code by Peter Meerwald.
Forcing all mute changes to go through set_mute() makes it easier to
check where the muted field is changed, and it also allows us to have
only one place where notifications for changed mute are sent.
This refactoring reduces duplication, as mute_changed() used to do the
same things as set_mute(). Other benefits are improved logging
(set_mute() logs the mute change, mute_changed() used to not do that)
and the soft mute state is kept up to date, because set_mute() sends
the SET_MUTE message to the IO thread.
The set_mute_in_progress flag is an extra precaution for preventing
recursion in case a sink/source implementation's set_mute() callback
causes mute_changed() to be called. Currently there are no such
implementations, but I think that would be a valid thing to do, so
some day there might be such implementation.
Currently the alsa sink and source write directly to s->muted during
initialization, but I think it's better to avoid direct writes, and
use the set_mute() function instead, because that makes it easier to
figure out where s->muted is modified. This patch prevents the
set_mute() call from crashing in the state assertion.
