idea is to allow optimized code path (similar to volume code)
and rework/specialize mixing cases to enable runtime performance improvements
no functionality changes in this patch
Signed-off-by: Peter Meerwald <pmeerw@pmeerw.net>
The patch intends to reduce computational load when resampling AND remapping. The PA
resampler performs the following steps:
sample format conversion -> remapping -> resampling -> sample format conversion
In case the number of output channels is higher than the number of input channels, the
resampler has to be run more often than necessary. E.g. in case of mono to 4-channel remapping,
the resampler runs on 4 channels separately.
To ímprove this, the PA resampler pipeline is made adaptive:
if out-channels <= in-channels:
sample format conversion -> remapping -> resampling -> sample format conversion
if out-channels > in-channels:
sample format conversion -> resampling -> remapping -> sample format conversion
Signed-off-by: Peter Meerwald <p.meerwald@bct-electronic.com>
Initialize the variable to zero by using pa_xnew0() instead of
pa_xnew(). This also allows us to remove a bunch of other zero
initialization statements.
Reported-by: Peter Meerwald <p.meerwald@bct-electronic.com>
bug probably caused by alignment requirement; sizeof(a->w) is a pointer, sizeof(a->w_arr) is an array
Signed-off-by: Peter Meerwald <p.meerwald@bct-electronic.com>
computes EC block size in frames (rounded down to nearest power-of-2) based
on sample rate and milliseconds
move code from speex AEC implementation to module-echo-cancel such that
functionality can be reused by other AEC implementations
Signed-off-by: Peter Meerwald <p.meerwald@bct-electronic.com>
prevents
CC module_ladspa_sink_la-module-ladspa-sink.lo
modules/module-ladspa-sink.c:1332:5: warning: "HAVE_DBUS" is not defined
modules/module-ladspa-sink.c:1370:5: warning: "HAVE_DBUS" is not defined
in case HAVE_DBUS is not available
Signed-off-by: Peter Meerwald <pmeerw@pmeerw.net>
alsa/use-case.h in needed
require at least version 1.0.24 in configure.ac
prevents the following error at compile time:
CC libalsa_util_la-alsa-util.lo
In file included from modules/alsa/alsa-mixer.h:51,
from modules/alsa/alsa-util.h:36,
from modules/alsa/alsa-util.c:46:
modules/alsa/alsa-ucm.h:27:22: error: use-case.h: No such file or directory
In file included from modules/alsa/alsa-mixer.h:51,
from modules/alsa/alsa-util.h:36,
from modules/alsa/alsa-util.c:46:
modules/alsa/alsa-ucm.h:89: error: expected ‘)’ before ‘*’ token
modules/alsa/alsa-ucm.h:169: error: expected specifier-qualifier-list before ‘snd_use_case_mgr_t’
make[3]: *** [libalsa_util_la-alsa-util.lo] Error 1
Signed-off-by: Peter Meerwald <pmeerw@pmeerw.net>
Remixing one channel map to another is (except for special cases) done
via a linear mapping between channels, whose corresponding matrix is
computed by calc_map_table(). The k-th row in this matrix corresponds to
the coefficients of the linear combination of the input channels that
result in the k-th output channel. In order to avoid clipping of samples
we require that the sum of these coefficients is (at most) 1. This
commit ensures this.
Prior to this commit tests/remix-test.c gives 52 of 132 matrices that
violate this property. For example:
'front-left,front-right,front-center,lfe' -> 'front-left,front-right'
prior this commit after this commit
I00 I01 I02 I03 I00 I01 I02 I03
+------------------------ +------------------------
O00 | 0.750 0.000 0.375 0.375 O00 | 0.533 0.000 0.267 0.200
O01 | 0.000 0.750 0.375 0.375 O01 | 0.000 0.533 0.267 0.200
Building the matrix is done in several steps. However, only insufficient
measures are taken in order to preserve a row-sum of 1.0 (or leaves it
at 0.0) after each step. The current patch adds a post-processing step
in order check for each row whether the sum exceeds 1.0 and, if
necessary, normalizes this row. This allows for further simplifactions:
- The insufficient normalizations after some steps are removed. Gains
are adapted to (partially) resemble the old matrices.
- Handling unconnected input channls becomes a lot simpler.
- Separate the cases with PA_RESAMPLER_NO_REMAP or PA_RESAMPLER_NO_REMIX
set and remove redundant if-conditions.
- Fix C90 compiler warning due to mixing code and variable declaration.
- Do not repeatedly count number of left, right and center channels in
the input channel map.
The logic of calc_map_table() remains unaltered.
Capability dropping when changing the user in the system
mode was previously implemented by calling
prctl(PR_SET_KEEPCAPS, 0), but that doesn't necessarily
work. It's possible that the KEEPCAPS flag is locked to 1,
in which case the prctl() call fails with EPERM (this
happens at least on Harmattan). This patch implements
explicit capability dropping after changing the user.
Earlier, -1 was returned if the memchunk size was not a multiple of the frame
size. Now, it is verified unconditionally through an assertion. Error code -1
is still returned when the memblock queue is full.
In those few cases where the return value of pa_memblockq_push() is checked,
an overflow is assumed to be the reason in case an error code is returned.
If 'PlaybackChannels' and 'CaptureChannels' are absent in the UCM
file for a device, assume the device is stereo duplex.
Reported-by: Luke Yelavich <luke.yelavich@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Now you can actually see *which* sink/source that sends a specific
message to the log, which is quite useful if you have more than
one sound card.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
pa_silence_memory() pulls sample-util as a dependency, so it had to
be moved from libpulsecore to libpulsecommon. sample-util in turn
pulls some more stuff.
use (1<<15) instead of 0x7fff as a factor when converting from s16 to float32
use (1<<31) instead of 0x7fffffff as a factor when converting from s32 to float32
the change is motivated by the following desireable properties:
* s16_from_f32(f32_from_s16(x)) == x for all possible s16 values
* x / (1.0f << 15) == x * (1.0f / (1 << 15)) for all x in s16
above changes enable easier optimization while guaranteeing bit-exact results
further, other audio sample conversion code (libavresample) does it the same way
v3 (comments Tanu):
* fix saturation in pa_sconv_s16le_from_f32ne_neon(), use vqrshrn
v2 (comments Tanu):
* fix comments in ARM NEON code
* use llrintf() in pa_sconv_s32le_from_float32ne()
Signed-off-by: Peter Meerwald <p.meerwald@bct-electronic.com>
Cc: Tanu Kaskinen <tanuk@iki.fi>
I went to implement the possibility to use the default sink/source
but found that it was already working. So I figured I'd update
the help text instead.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Problem: s16 to s32 conversion is performed as s16->float->s32 (via work
format float) for resamplers TRIVIAL, COPY, PEAKS.
Precision and efficiency suffers: e.g. 0x9fff results in 0x9ffe4001 (instead
of 0x9fff0000) and there are two sample format conversions instead of one
conversion.
Solution: If input or output format is s16, then choose the work format
to be s16 as well.
If remapping is to be performed, we could stick to work format float32ne for
precision reseans. This is debateable.
Signed-off-by: Peter Meerwald <p.meerwald@bct-electronic.com>
v2 (comments by Paul Menzel):
* generate test samples from -1..1, -0x8000..0x7fff
* check all output samples (not just half of them)
the idea is to compare the output of the C (reference) implementation
against the output of the optimized code path; currently, there are MMX
and SSE implementation for the mono-to-stereo remapper for s16 and float
sample formats
Signed-off-by: Peter Meerwald <p.meerwald@bct-electronic.com>
Cc: Paul Menzel <paulepanter@users.sourceforge.net>
SSE sconv was not tested before, only SSE2 was (on CPUs supporting both
instruction sets)
now both code path are tested on CPUs supporting both
Signed-off-by: Peter Meerwald <p.meerwald@bct-electronic.com>
nsamples should be forced to be a multiple of channels; do so correctly
and don't make nsamples larger than it actually is
Signed-off-by: Peter Meerwald <p.meerwald@bct-electronic.com>
This adds two new commands to pactl:
set-default-sink
set-default-source
This command has been part of the native protocol for a long time,
no reason not to expose it in pactl.
The use of the pseudo-blocking D-Bus calls leads to the problem that
NameLost signals are received after the reply to ReleaseName().
The problem with this is that a later acquisition of the same audio
device can potentially receive the NameLost signal corresponding to
the previous instance, due to the fact that the signal hasn't been
popped from the D-Bus message queue.
The simplest approach to solve this problem is to poll the actual name
owner from the D-Bus daemon, in order to make sure that we did really
lose the name.
The proposal uses a blocking call to GetNameOwner to avoid incosistent
states in the internal APIs: it would otherwise be possible to have a
"busy" device before the reservation has been lost, in the unlikely
case if some other process acquires the name before we got the
confirmation that the NameLost was actually true.
The function is interesting for both rd_device and rd_monitor so make
it part of the rd_device public API to avoid duplicated code.
The decision to move the function to reserve.c is motivated by the fact
that other projects (i.e. jack) use reserve.c only. Therefore, adding a
reserve->reserve-monitor dependency should be avoided.
