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>
"i->save_muted = i->save_muted || mute" makes no sense. The intention
was most likely to use "save" instead of "mute" in the assignment.
This line originates from reverting the volume ramping code, commit
8401572fd5.
The idea of "i->save_muted |= save" is that even if the mute state
doesn't change, save_muted should still be updated, but only if the
transition is from "don't save" to "save".
Changing "!i->muted == !mute" to "mute == i->muted" is cosmetic only.
The rationale behind the old form was probably that when we still had
pa_bool_t, booleans could in theory be defined as int, so comparing
the values without the ! operator was not entirely safe. That's
unnecessary now that we use the standard bool type, which can only
have values 0 or 1.
A value of 0 for adjust_time should disable rate adjustment.
Fix a bug where a 0 value causes rate adjustment to be called
continuously instead after an unsuspend event.
Initially (in commit ef422fa4ae),
pa_make_secure_dir followed a simple principle: "make a directory, or,
if it exists, check that it is suitable". Later this evolved into "make
a directory, or, if it exists, ensure that it is suitable". But the
check remained.
The check is now neither sufficient nor necessary. On POSIX-compliant
systems, the fstat results being checked are actually post-conditions of
fchmod and fchown. And on systems implementing POSIX ACLs, fstat only
reflects a part of the information relevant to the security of the
directory permissions, so PulseAudio could accept an existing insecure
directory anyway.
Also, the check still fires on non-POSIX-compliant filesystems like CIFS.
As a user cannot do anything to fix it, just accept insecure permissions
in this case.
