On FIONREAD returning 0 bytes, we cannot return success, as the caller
(rtpoll_work_cb in module-rtp-recv.c) would then try to
pa_memblock_unref(chunk.memblock) and, because memblock is NULL, trigger
an assertion.
Also we have to read out the possible empty packet from the socket, so
that the kernel doesn't tell us again and again about it.
Signed-off-by: Alexander E. Patrakov <patrakov@gmail.com>
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
Add the output from its sink-input attached callback and remove it
again from the detach callback. This simplifies some output_enable
and we can also avoid posting 2 messages for the sink.
Surround 2.1 is one of the more common surround profiles these days,
so it's about time we support it.
The "surround21" was added to alsa-lib a few months ago, and there
hasn't yet been an alsa-lib release since, but I doubt it will change.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
This name is more acurate with regards of what role we're currently
playing and we've already been using it in
pa_bluetooth_profile_to_string() since 449d6cb.
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.
The warnings were produced because the command-line flag redefined the
value of _FORTIFY_SOURCE coming from the specs on some distributions,
including Gentoo. So, undefine this macro before defining it.
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.
The callback just called pa_source_output_get_mute(), which doesn't
have any side effects, and the return value wasn't used either, so
the callback was essentially a no-op.
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.
Forcing all volume changes to go through set_volume_direct() makes
it easier to check where the stream volume is changed, and it also
allows us to have only one place where notifications for changed
volume are sent.
Forcing all reference volume changes to go through
set_reference_volume_direct() makes it easier to check where the
reference volume is changed, and it also allows us to have only one
place where notifications for changed reference volume are sent.
State can be used by remap function implementations to
speed up the remapping, e.g. by precomputing things or
even by generating specialized code for a specific channel
remapping task
Signed-off-by: Peter Meerwald <pmeerw@pmeerw.net>
Initialization of the remap structure now happens in one place
Rename calc_map_table() to setup_remap(), copy sample format and
channel specs; the remap structure is initialized when we know the
work sample format of the resampler
Signed-off-by: Peter Meerwald <pmeerw@pmeerw.net>
pa_init_remap_func() only sets the appropriate remapping function, it
does not initialize the pa_remap struct
Signed-off-by: Peter Meerwald <pmeerw@pmeerw.net>
