Add a user defined parameter lfe-crossover-freq for the lfe-filter,
to pass this parameter to the lfe-filter, we need to change the
pa_resampler_new() API as well.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
When enable-lfe-remixing is set, an LFE channel is present in the
resampler's destination channel map but not in the source channel map,
we insert a low-pass filter instead of just averaging the channels.
Other channels will get a high-pass filter.
In this patch, the crossover frequency is hardcoded to 120Hz (to be fixed
in later patches).
Note that in current state the LFE filter is
- not very optimised
- not rewind friendly (rewinding can cause audible artifacts)
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Added ID and names for the resampler presets and also updated the working sample rate deduction to take the new resampler into account. The initial libsoxr backend version does not variable rate resampling, so it is disabled in this case.
FSF addresses used in PA sources are no longer valid and rpmlint
generates numerous warnings during packaging because of this.
This patch changes all FSF addresses to FSF web page according to
the GPL how-to: https://www.gnu.org/licenses/gpl-howto.en.html
Done automatically by sed-ing through sources.
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.
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>
Reintroduces a cleaned-up version of commit 30ce3a14e5 which
was reverted by 1ce71cbd82; for more information see
http://thread.gmane.org/gmane.comp.audio.pulseaudio.general/17479/focus=17487
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>
the patch changes the interface of the (internal) fit_buf() function:
fit_buf() manages the memblock of the buf chunk, it reallocates the memblock
if the requested number of bytes ('len') if larger than the memblock's size ('size')
and optionally preserves 'copy' bytes
the code should be in line with the comment now
Signed-off-by: Peter Meerwald <pmeerw@pmeerw.net>
I think this makes the code a bit nicer to read and write. This also
reduces the chances of off-by-one errors when checking the bounds of
sample rate values.
I think this makes the code a bit nicer to read and write. This also
reduces the chances of off-by-one errors when checking the bounds of
the sample format value.
buf in struct ffmpeg_data is reset() initially and freed, but never
actually used
when a new block is allocated ffmpeg_data->buf[c].length is used
(which is always 0) to compute the new block size
so, drop buf
Signed-off-by: Peter Meerwald <p.meerwald@bct-electronic.com>
leftover_buf points to the output buffer of a stage containing leftover
data; similar for leftover_buf_size and have_leftover
Signed-off-by: Peter Meerwald <pmeerw@pmeerw.net>
some resampler implementations (e.g. libsamplerate and ffmpeg) do not consume
the entire input buffer; the impl_resample() function now has a return value
returning the number of frames in the input buffer not processed
these frames must be saved in appropriate buffer and presented together with
new input data
also change the parameter names from in_samples, out_samples to in_n_frames,
out_n_frames, respectively (n_frames = samples / channels)
Signed-off-by: Peter Meerwald <pmeerw@pmeerw.net>
This patch fixes a small mistake where we actually log that we are
reverting to the auto resampler if we can't use the 'copy' resampler but
never do the revert.
This would lead to a crash if the user chooses the 'copy' resampler and
then tries to play something that needs to be resampled.
With very low input sample rates the memory pool max block size may
not be big enough, in which case we should return the size of one
frame. Returning zero caused crashing.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=68616
This adds asserts to check if the implementation has an update rate
function defined for the unlikely event that some implementation forgets
to assign a update rate function we can simply bail.
It is expected from the resampling implementations to have such a
function even if the state of the resampler is completely reset.
This patch fixes this assertion:
Assertion 'r->i_ss.rate >= r->o_ss.rate' failed at ../../src/pulsecore/resampler.c:1744, function peaks_init(). Aborting.
The pa_resampler struct contains many implementation specific
structures. These create overhead and don't belong there anyways.
This patch moves the implementation specific structures out of the
pa_resampler structure.
This consumes less power, has low (no?) perceivable difference, and
allows the default configuration to work out of the box on low-end
systems (such as netbooks).
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>
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.
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>
