PDL-Audio/audio.pd

$VERSION = '1.1';

pp_setversion $VERSION;
pp_beginwrap (); # force error with older PPs

pp_addpm {At => Top}, <<'EOD';
=head1 NAME

PDL::Audio - Some PDL functions intended for audio processing.

=head1 SYNOPSIS

  use PDL;
  use PDL::Audio;

=head1 DESCRIPTION

Oh well ;) Not much "introductory documentation" has been written yet :(

=head2 NOTATION

Brackets around parameters indicate that the respective parameter is
optional and will be replaced with some default value when absent (or
C<undef>, which might be different in other packages).

The sampling frequency and duration are by default (see individual
descriptions) given in cycles/sample (or samples in case of a
duration). That means if you want to specify a duration of two seconds,
you have to multiply by the sampling frequency in HZ, and if you want
to specify a frequency of 440 Hz, you have to divide by the sampling
frequency:

 # Syntax: gen_oscil duration*, frequency/
 $signal = gen_oscil 2*HZ, 440/HZ;
 # with a sampling frequency of 44100 Hertz:
 $signal = gen_oscil 2*44100, 440/44100;

To help you, the required unit is given as a type suffix in the parameter
name. A "/" means that you have to divide by the sampling frequency (to
convert from Hertz) and a suffix of "*" indicates that a multiplication is
required.

Most parameters named "size", "duration" (or marked with "*") can
be replaced by a piddle, which is then used to give length and from
(mono/stereo).

=head2 HEADER ATTRIBUTES

The following header attributes are stored and evaluated by most
functions. PDL::Audio provides mutator methods for all them (e.g.
 
 print "samplerate is ", $pdl->rate;
 $pdl->comment("set the comment to this string");

=over 4

=item rate

The sampling rate in hz.

=item filetype

The filetype (wav, au etc..). Must be one of:

  FILE_NEXT FILE_AIFC FILE_RIFF FILE_BICSF FILE_NIST FILE_INRS FILE_ESPS
  FILE_SVX FILE_VOC FILE_SNDT FILE_RAW FILE_SMP FILE_SD2 FILE_AVR
  FILE_IRCAM FILE_SD1 FILE_SPPACK FILE_MUS10 FILE_HCOM FILE_PSION
  FILE_MAUD FILE_IEEE FILE_DESKMATE FILE_DESKMATE_2500 FILE_MATLAB
  FILE_ADC FILE_SOUNDEDIT FILE_SOUNDEDIT_16 FILE_DVSM FILE_MIDI
  FILE_ESIGNAL FILE_SOUNDFONT FILE_GRAVIS FILE_COMDISCO FILE_GOLDWAVE
  FILE_SRFS FILE_MIDI_SAMPLE_DUMP FILE_DIAMONDWARE FILE_REALAUDIO
  FILE_ADF FILE_SBSTUDIOII FILE_DELUSION FILE_FARANDOLE FILE_SAMPLE_DUMP
  FILE_ULTRATRACKER FILE_YAMAHA_SY85 FILE_YAMAHA_TX16 FILE_DIGIPLAYER
  FILE_COVOX FILE_SPL FILE_AVI FILE_OMF FILE_QUICKTIME FILE_ASF
  FILE_YAMAHA_SY99 FILE_KURZWEIL_2000 FILE_AIFF FILE_AU

=item path

The filename (or file specification) used to load or save a file.

=item format

Specifies the type the underlying file format uses. The samples will always be in short or long signed format.

Must be one of

   FORMAT_NO_SND FORMAT_16_LINEAR FORMAT_8_MULAW FORMAT_8_LINEAR
   FORMAT_32_FLOAT FORMAT_32_LINEAR FORMAT_8_ALAW FORMAT_8_UNSIGNED
   FORMAT_24_LINEAR FORMAT_64_DOUBLE FORMAT_16_LINEAR_LITTLE_ENDIAN
   FORMAT_32_LINEAR_LITTLE_ENDIAN FORMAT_32_FLOAT_LITTLE_ENDIAN
   FORMAT_64_DOUBLE_LITTLE_ENDIAN FORMAT_16_UNSIGNED
   FORMAT_16_UNSIGNED_LITTLE_ENDIAN FORMAT_24_LINEAR_LITTLE_ENDIAN
   FORMAT_32_VAX_FLOAT FORMAT_12_LINEAR FORMAT_12_LINEAR_LITTLE_ENDIAN
   FORMAT_12_UNSIGNED FORMAT_12_UNSIGNED_LITTLE_ENDIAN COMPATIBLE_FORMAT

PDL::Audio conviniently defines the following aliases for the following
constants, that are already correct for the host byteorder:

   FORMAT_ULAW_BYTE FORMAT_ALAW_BYTE FORMAT_LINEAR_BYTE
   FORMAT_LINEAR_SHORT FORMAT_LINEAR_USHORT FORMAT_LINEAR_LONG
   FORMAT_LINEAR_FLOAT FORMAT_LINEAR_DOUBLE

=item comment

The file comment (if any).

=item device

The device to output audio. One of:

   DEV_DEFAULT DEV_READ_WRITE DEV_ADAT_IN DEV_AES_IN DEV_LINE_OUT
   DEV_LINE_IN DEV_MICROPHONE DEV_SPEAKERS DEV_DIGITAL_IN DEV_DIGITAL_OUT
   DEV_DAC_OUT DEV_ADAT_OUT DEV_AES_OUT DEV_DAC_FILTER DEV_MIXER
   DEV_LINE1 DEV_LINE2 DEV_LINE3 DEV_AUX_INPUT DEV_CD_IN DEV_AUX_OUTPUT
   DEV_SPDIF_IN DEV_SPDIF_OUT
   
=back

=head2 EXPORTED CONSTANTS

In addition to the exported constants described above (and later in
the function descriptions), this module also exports the mathematical
constants M_PI and M_2PI, so watch out for clashes!

=cut
EOD

@cconsts = qw(
   SNDLIB_DEFAULT_DEVICE SNDLIB_READ_WRITE_DEVICE
   SNDLIB_ADAT_IN_DEVICE SNDLIB_AES_IN_DEVICE
   SNDLIB_LINE_OUT_DEVICE SNDLIB_LINE_IN_DEVICE
   SNDLIB_MICROPHONE_DEVICE SNDLIB_SPEAKERS_DEVICE
   SNDLIB_DIGITAL_IN_DEVICE SNDLIB_DIGITAL_OUT_DEVICE
   SNDLIB_DAC_OUT_DEVICE SNDLIB_ADAT_OUT_DEVICE
   SNDLIB_AES_OUT_DEVICE SNDLIB_DAC_FILTER_DEVICE
   SNDLIB_MIXER_DEVICE SNDLIB_LINE1_DEVICE
   SNDLIB_LINE2_DEVICE SNDLIB_LINE3_DEVICE
   SNDLIB_AUX_INPUT_DEVICE SNDLIB_CD_IN_DEVICE
   SNDLIB_AUX_OUTPUT_DEVICE SNDLIB_SPDIF_IN_DEVICE
   SNDLIB_SPDIF_OUT_DEVICE

   SNDLIB_NO_SND SNDLIB_16_LINEAR SNDLIB_8_MULAW SNDLIB_8_LINEAR
   SNDLIB_32_FLOAT SNDLIB_32_LINEAR SNDLIB_8_ALAW SNDLIB_8_UNSIGNED
   SNDLIB_24_LINEAR SNDLIB_64_DOUBLE SNDLIB_16_LINEAR_LITTLE_ENDIAN
   SNDLIB_32_LINEAR_LITTLE_ENDIAN SNDLIB_32_FLOAT_LITTLE_ENDIAN
   SNDLIB_64_DOUBLE_LITTLE_ENDIAN SNDLIB_16_UNSIGNED
   SNDLIB_16_UNSIGNED_LITTLE_ENDIAN SNDLIB_24_LINEAR_LITTLE_ENDIAN
   SNDLIB_32_VAX_FLOAT SNDLIB_12_LINEAR SNDLIB_12_LINEAR_LITTLE_ENDIAN
   SNDLIB_12_UNSIGNED SNDLIB_12_UNSIGNED_LITTLE_ENDIAN
   SNDLIB_COMPATIBLE_FORMAT

   NeXT_sound_file AIFC_sound_file RIFF_sound_file
   BICSF_sound_file NIST_sound_file INRS_sound_file
   ESPS_sound_file SVX_sound_file VOC_sound_file
   SNDT_sound_file raw_sound_file SMP_sound_file
   SD2_sound_file AVR_sound_file IRCAM_sound_file
   SD1_sound_file SPPACK_sound_file MUS10_sound_file
   HCOM_sound_file PSION_sound_file MAUD_sound_file
   IEEE_sound_file DeskMate_sound_file DeskMate_2500_sound_file
   Matlab_sound_file ADC_sound_file SoundEdit_sound_file
   SoundEdit_16_sound_file DVSM_sound_file MIDI_file
   Esignal_file soundfont_sound_file gravis_sound_file
   comdisco_sound_file goldwave_sound_file srfs_sound_file
   MIDI_sample_dump DiamondWare_sound_file RealAudio_sound_file
   ADF_sound_file SBStudioII_sound_file Delusion_sound_file
   Farandole_sound_file Sample_dump_sound_file Ultratracker_sound_file
   Yamaha_SY85_sound_file Yamaha_TX16_sound_file digiplayer_sound_file
   Covox_sound_file SPL_sound_file AVI_sound_file
   OMF_sound_file Quicktime_sound_file asf_sound_file
   Yamaha_SY99_sound_file Kurzweil_2000_sound_file AIFF_sound_file

   BANDPASS DIFFERENTIATOR HILBERT
);

for (@cconsts) {
   local $_ = $_;
   s/^SNDLIB_(.*)_DEVICE/DEV_$1/;
   s/^SNDLIB_/FORMAT_/;
   s/MIDI_file/FILE_MIDI/;
   s/MIDI_sample_dump/FILE_MIDI_sample_dump/;
   s/Esignal_file/FILE_Esignal/;
   s/(.*)_sound_file/FILE_$1/;
   push @pconsts, uc $_;
}

@METHODS = qw(
   waudio cut_leading_silence cut_trailing_silence cut_silence filter_zpolar
   partials2polynomial scale2short filter_fir filter_iir rfft irfft spectrum
   filter_comb filter_notch filter_allpass filter_src filter_granulate
   filter_contrast_enhance describe_audio filter_center ring_modulate
   amplitude_modulate filter_ppolar design_remez_fir concat
   gen_oscil gen_sawtooth gen_square gen_triangle gen_asymmetric_fm gen_rand_1f
   gen_env audiomix gen_sum_of_cosines gen_sine_summation gen_fft_window
   gen_pulse_train gen_rand gen_from_table gen_from_partials partials2waveshape

);
@EXPORT = (qw(
   sound_format_name sound_type_name raudio
   FORMAT_ULAW_BYTE FORMAT_ALAW_BYTE FORMAT_LINEAR_BYTE
   FORMAT_LINEAR_SHORT FORMAT_LINEAR_USHORT FORMAT_LINEAR_LONG
   FORMAT_LINEAR_FLOAT FORMAT_LINEAR_DOUBLE
   M_PI M_2PI
), @METHODS, @pconsts);

for (@EXPORT) {
   pp_add_exported "", $_;
}

pp_addpm "\@METHODS = qw(@METHODS);\n";

$addboot = "{\nHV *stash = gv_stashpvn(\"PDL::Audio\", 10, TRUE);";

for (0..$#cconsts) {
   $addboot .= "   newCONSTSUB(stash, \"$pconsts[$_]\", newSViv($cconsts[$_]));\n";
}
$addboot .= "   newCONSTSUB(stash, \"M_PI\" , newSVnv(M_PI ));\n";
$addboot .= "   newCONSTSUB(stash, \"M_2PI\", newSVnv(M_2PI));\n";
$addboot .= '}
   mus_set_error_handler (mus_barfer);
';

pp_add_boot $addboot;

pp_addpm <<'EOD';

use PDL::LiteF;
use PDL::Complex;
use Carp;

$bigendian = 0x55 == unpack "S", "\0U";

sub FORMAT_ULAW_BYTE    (){ FORMAT_8_MULAW }
sub FORMAT_ALAW_BYTE    (){ FORMAT_8_ALAW  }
sub FORMAT_LINEAR_BYTE  (){ FORMAT_8_UNSIGNED }
sub FORMAT_LINEAR_SHORT (){ $bigendian ? FORMAT_16_LINEAR   : FORMAT_16_LINEAR_LITTLE_ENDIAN   }
sub FORMAT_LINEAR_USHORT(){ $bigendian ? FORMAT_16_UNSIGNED : FORMAT_16_UNSIGNED_LITTLE_ENDIAN }
sub FORMAT_LINEAR_LONG  (){ $bigendian ? FORMAT_32_LINEAR   : FORMAT_32_LINEAR_LITTLE_ENDIAN   }
sub FORMAT_LINEAR_FLOAT (){ $bigendian ? FORMAT_32_FLOAT    : FORMAT_32_FLOAT_LITTLE_ENDIAN    }
sub FORMAT_LINEAR_DOUBLE(){ $bigendian ? FORMAT_32_DOUBLE   : FORMAT_32_DOUBLE_LITTLE_ENDIAN   }

sub FILE_AU             (){ FILE_NEXT }

# provide some accessor methods
for my $field (qw(rate comment filetype path format)) {
   *{"PDL::$field"} = sub {
      my $pdl = shift;
      my $hdr = $pdl->gethdr;
      if (@_) {
         $hdr->{$field} = $_[0];
         $pdl->sethdr($hdr);
      };
      $hdr->{$field};
   };
}

=head2 sound_format_name format_code

Return the human-readable name of the file format with code C<format_code>.

=head2 sound_type_name type_code

Return the human-readable name of the sample type with code C<type_code>.

=head2 describe_audio piddle

Describe the audio stream contained in piddle and return it as a string. A
fresh piddle might return:

 mono sound with 27411 samples

Whereas a freshly loaded soundfile might yield:

 stereo sound with 27411 samples, original name "kongas.wav", type 2 (RIFF),
 rate 11025/s (duration 2.49s), format 7 (8-bit unsigned)

=cut

sub describe_audio($) {
   my $pdl = shift;
   my ($channels, $samples) = $pdl->dims;
   my $chan = $channels  < 2 ? "mono" :
              $channels == 2 ? "stereo" :
              $channels == 4 ? "quad channel" :
                               "$channels channel";
   my $desc = "$chan sound with $samples samples";
   $desc .= sprintf ", original name \"%s\"", $pdl->path if $pdl->path;
   $desc .= sprintf ", type %d (%s)", $pdl->filetype, sound_type_name($pdl->filetype) if $pdl->filetype;
   $desc .= sprintf ", rate %d/s (duration %.2fs)", $pdl->rate, $samples/$pdl->rate if $pdl->rate;
   $desc .= sprintf ", format %d (%s)", $pdl->format, sound_format_name($pdl->format) if $pdl->format;
   $desc
}

=head2 raudio path, [option-hash], option => value, ...

Reads audio data into the piddle. Options can be anything, most useful
values are C<filetype>, C<rate>, C<channels> and C<format>. The returned
piddle is represents "time" in the outer dimension, and samples in the
inner (i.e. scalars for mono files, 2-vectors for stereo files):
   
 [ [left0, right0], [left1, right1], [left2, right2], ...]

 # read any file
 $pdl = raudio "file.wav";
 # read a file. if it is a raw file preset values
 $pdl = raudio "file.raw", filetype => FILE_RAW, rate => 44100, channels => 2;

=head2 waudio pdl, [option-hash], option => value, ...

Writes a pdl as a file. See L<raudio> for options and format. The path and
other metadata is taken from the header, whcih cna be overwritten using
options, e.g.:

 # write a file, using the header of another piddle
 $pdl->waudio ($orig_file->gethdr);
 # write pdl as .au file, take rate from the header
 $pdl->waudio (path => "piddle.au", filetype => FILE_AU, format => FORMAT_16_LINEAR;

=cut

# read a sound file
sub raudio {
   my $path = shift;
   my %hdr = (); %hdr = (%hdr, %{+shift}) if ref $_[0]; %hdr = (%hdr, @_) if @_;

   # for raw files this is necessary
   mus_set_raw_header_defaults $hdr{rate}    || 44100,
                               $hdr{channels}|| 1,
                               $hdr{format}  || FORMAT_LINEAR_SHORT;

   $hdr{path}       = $path;
   $hdr{filetype}   = sound_header_type $path;
                         $hdr{filetype} >= 0 or barf "$path: ".audio_error_name audio_error;
   $hdr{rate}       = sound_srate $path;
   $hdr{comment}    = sound_comment $path if defined sound_comment $path;
   $hdr{format}     = sound_data_format $path;
   my $channels     = sound_chans $path;
   my $frames       = sound_frames $path;

   my $fd = open_sound_input $path; $fd >= 0 or barf "$path: ".audio_error_name audio_error;

   my $pdl = zeroes long, $frames, $channels;
   $pdl->clump(-1)->read_sound ($fd, $channels, $frames)
                                        >= 0 or barf "$path: ".audio_error_name audio_error;
   (close_sound_input $fd)              >= 0 or barf "$path: ".audio_error_name audio_error;
   $pdl = $pdl->short->xchg(0,1);
   $pdl = $pdl->clump(2) if $channels == 1;
   $pdl->sever;
   $pdl->sethdr(\%hdr);
   $pdl
}

sub _audio_make_plain {
   my $pdl = shift;
   if ($pdl->getndims == 1) {
      ($pdl, 1, $pdl->getdim(0))
   } else {
      ($pdl->xchg(0,1)->clump(-1), $pdl->dims)
   }
}

sub waudio {
   my $pdl = shift;
   my %hdr = %{$pdl->gethdr || {}}; %hdr = (%hdr, %{+shift}) if ref $_[0]; %hdr = (%hdr, @_) if @_;
   my ($channels, $frames);

   $hdr{filetype}   = FILE_NEXT unless exists $hdr{filetype};
   $hdr{format}   ||= FORMAT_16_LINEAR;
   $hdr{rate}     ||= 44100;

   ($pdl, $channels, $frames) = _audio_make_plain $pdl->convert(long);

   1 <= $channels && $channels <= 2
      or croak "can only write mono or stereo (one or two channel) files, not $channels channel files";

   my $fd = open_sound_output $hdr{path}, $hdr{rate}, $channels, $hdr{format}, $hdr{filetype}, $hdr{comment};
   $fd >= 0 or barf "$hdr{$path}: ".audio_error_name audio_error;
   $pdl->clump(-1)->write_sound($fd, $channels, $frames)
       >= 0 or barf "$path: ".audio_error_name audio_error;
   (close_sound_output $fd, mus_samples2bytes $hdr{format}, $frames * $channels)
       >= 0 or barf "$hdr{$path}: ".audio_error_name audio_error;
}

=head2 cut_leading_silence pdl, level

Cuts the leading silence (i.e. all samples with absolute value < level)
and returns the resulting part.

=head2 cut_trailing_silence pdl, level

Cuts the trailing silence.

=head2 cut_silence pdl, level

Calls C<cut_leading_silence> and C<cut_trailing_silence> and returns the
result.

=cut

sub cut_leading_silence {
   my $pdl = shift;
   my $level = 1*shift;
   my $skip = which (abs($pdl) > $level);
   $skip = $skip->nelem ? $skip->at(0) : 0;
   $pdl->slice("$skip:-1")
}

sub cut_trailing_silence {
   my $pdl = shift;
   my $level = 1*shift;
   $level = 400000;
   my $skip = which (abs($pdl) > $level);
   $skip = $skip->nelem ? $skip->at(-1) : -1;
   $skip-- if $skip > 0;
   $pdl->slice("0:$skip")
}

sub cut_silence {
   $_[0]->cut_leading_silence($_[1])
        ->cut_trailing_silence($_[1])
}

# have we been a bad boy?

for (@METHODS) {
   *{"PDL::$_"} = \&$_;
}
EOD

for my $d (qw(read write)) {
   pp_def($d.'_sound',
          Pars => ($d eq "read" ? '[o]' : '') . 'sample(n)',
          OtherPars => "int fd; int chans; int frames",
          GenericTypes => [L],
          PMCode => '', 
          Doc => undef,
          Code => q@
              int chans = $COMP(chans);
              int frames = $COMP(frames);
              int **bufs = malloc (chans * sizeof (int *));
              int i;
              dSP;

              /* hmm... */
              assert (sizeof (PDL_long) == sizeof (int));

              for (i = 0; i < chans; i++)
                bufs[i] = (int *)$P(sample) + i * frames;

              XPUSHs (sv_2mortal (newSViv (@.$d.q@_sound ($COMP(fd), 0, frames - 1, chans, bufs))));

              free (bufs);
          @,
   );
}

pp_addhdr <<'EOH';
#include "sndlib/sndlib.h"
#include "xlib.h"

static void
mus_barfer (int err_type, char *err_msg)
{
  barf ("%s [MUSERR]", err_msg);
}

EOH

pp_addpm <<'EOP';

=head2 playaudio pdl, [option-hash], option => value ...

Play the piddle as an audio file. Options can be supplied either through
the option hash (a hash-reference), through the pdl header or the options:

 # play a piddle that has a valid header (e.g. from raudio)
 $pdl->playaudio;
 # play it with a different samplerate
 $pdl->playaudio(rate => 22050);

=cut

EOP

pp_def('playaudio',
       Pars => 'sample(n)',
       OtherPars => "int srate; int chans; int format; int dev; int bufsize",
       GenericTypes => [B,U,S,L],
       Doc => undef,
       PMCode => q!
           sub PDL::playaudio {
               my $pdl = shift;
               my %hdr = %{$pdl->gethdr || {}}; %hdr = (%hdr, %{+shift}) if ref $_[0]; %hdr = (%hdr, @_) if @_;
               my $chans;
               $hdr{format}  ||= FORMAT_LINEAR_SHORT;
               ($pdl, $chans) = _audio_make_plain $pdl;
               &PDL::_playaudio_int($pdl,
                  $hdr{rate}    || 44100,
                  $chans,
                  $hdr{format},
                  $hdr{device}  || DEV_DEFAULT,
                  $hdr{bufsize} || 32768);
           }
       !,
       Code => q@
           int ns;
           int line;

           if (sizeof ($GENERIC ()) != mus_format2bytes ($COMP(format)))
             barf ("pdl datatype and selected audio format do not match");

           if (initialize_audio () < 0)
             barf ("playaudio: %s", audio_error_name (audio_error ()));

           if ((line = open_audio_output ($COMP(dev), $COMP(srate), $COMP(chans), $COMP(format), $COMP(bufsize))) < 0)
             barf ("playaudio: %s", audio_error_name (audio_error ()));

           ns = $SIZE(n);
           threadloop %{
             if (write_audio (line, (void *)$P(sample), ns * sizeof ($GENERIC())))
               barf ("playaudio: %s", audio_error_name (audio_error ()));
           %}
           if (close_audio (line) < 0) 
             barf ("playaudio: %s", audio_error_name (audio_error ()));
           @,
);

pp_def 'ulaw2linear',
       Pars => 'byte u(n); short [o] s(n)',
       GenericTypes => [B],
       Doc => 'conversion from (m)u-law into signed, linear, 16 bit samples (rather slow)',
       Code => q! loop(n) %{ $s() = st_ulaw_to_linear ($u()); %} !,
;

pp_def 'linear2ulaw',
       Pars => 'short s(n); byte [o] u(n)',
       GenericTypes => [S],
       Doc => 'conversion from signed, linear, 16 bit samples into (m)u-law (rather slow)',
       Code => q! loop(n) %{ $u() = st_linear_to_ulaw ($s()); %} !,
;

pp_def 'alaw2linear',
       Pars => 'byte u(n); short [o] s(n)',
       GenericTypes => [B],
       Doc => 'conversion from A-law into signed, linear, 16 bit samples (rather slow)',
       Code => q! loop(n) %{ $s() = st_Alaw_to_linear ($u()); %} !,
;

pp_def 'linear2alaw',
       Pars => 'short s(n); byte [o] u(n)',
       GenericTypes => [S],
       Doc => 'conversion from signed, linear, 16 bit samples into A-law (rather slow)',
       Code => q! loop(n) %{ $u() = st_linear_to_Alaw ($s()); %} !,
;

pp_addpm <<'EOP';

=head2 gen_oscil duration*, freq/, phase-mod, [fm-mod/]

=head2 gen_sawtooth duration*, freq/, phase-mod, [fm-mod/]

=head2 gen_square duration*, freq/, phase-mod, duty, [fm-mod/]

=head2 gen_triangle duration*, freq/, phase-mod, [fm-mod/]

=head2 gen_pulse_train duration*, freq/, phase-mod, [fm-mod/]

=head2 gen_rand duration*, freq/

=head2 gen_rand_1f duration*

All of these functions generate appropriate waveforms with frequency
C<freq> (cycles/sample) and phase C<phase> (0..1).

The C<duration> might be either a piddle (which gives the form of the
output) or the number of samples to generate.

The output samples are between -1 and +1 (i.e. "-1 <= s <= +1").

The C<duty> parameter of the square generator influences the duty cycle of
the signal. Zero means 50%-50%, 0.5 means 75% on, 25% off, -0.8 means 10%
on, 90% off etc... Of course, the C<duty> parameter might also be a vector
of size C<duration>.

=cut

# return the time offset for duration, freq, phase, fm_mod
sub _dur2time($$;$$) {
   my ($dur, $freq, $phase, $fm_mod) = @_;
   zeroes($dur)->xvals*$freq+cumusumover($fm_mod)+$phase
}

sub gen_oscil($$;$$) {
   my ($dur, $freq, $phase, $fm_mod) = @_;
   sin M_2PI*_dur2time($dur,$freq,$phase,$fm_mod);
}

sub gen_sawtooth($$;$$) {
   my ($dur, $freq, $phase, $fm_mod) = @_;
   # maybe better use abs?
   $freq = $fm_mod * $freq if defined $fm_mod;
   (_dur2time($dur,$freq,$phase,$fm_mod) % 1 - 0.5) * 2;
}

sub gen_square($$;$$$) {
   my ($dur, $freq, $phase, $duty, $fm_mod) = @_;
   (gen_sawtooth($dur,$freq,$phase,$fm_mod) < $duty) * 2 - 1;
}

sub gen_triangle($$;$$) {
   my ($dur, $freq, $phase, $fm_mod) = @_;
   my $st = gen_sawtooth($dur,$freq,$phase,$fm_mod);
   $st * (($st < 0) * 2 - 1);
}

sub gen_pulse_train($$;$$) {
   my ($dur, $freq, $phase, $fm_mod) = @_;
   my $st = gen_sawtooth($dur,$freq,$phase,$fm_mod);
   $st < $st->rotate(1);
}

sub gen_rand($$) {
   my ($dur, $freq) = @_;
   my $z = zeroes $dur;
   my $r = (random zeroes $freq * $z->getdim(0) + 1) * 2 - 1;
   $r->index($freq * $z->xvals)->sever;
}

sub gen_rand_1f($) {
   my ($dur) = @_;
   $dur = zeroes $dur;
   _gen_noise_1f($dur);
   $dur;
}

=head2 gen_env duration*, xvals, yvals, [base]

Generates an interpolated envelope between the points given by xvals and
yvals.  When base == 1 (the default) then the values will be linearly
interpolated, otherwise they follow an exponential curve that is bend
inwards (base < 1) or outwards (base > 1).

 # generate a linear envelope with attack in the first 10%
 gen_env 5000, [0 1 2 9 10], [0 1 0.6 0.6 0];

=cut

sub gen_env($;@) {
   my ($dur, $x, $y, $base) = @_;
   my $pdl = zeroes $dur;
   $base ||= 1;
   $base == 1 or barf "base values != 1 are not yet implemented\n";
   my @x = ref $x eq "ARRAY" ? @$x : $x->list;
   my @y = ref $y eq "ARRAY" ? @$y : $y->list;
   my $f = $pdl->getdim(0) / $x[-1];
   @x == @y or barf "gen_env: x and y lists have different sizes";
   my $x0 = 0;
   my $y0 = 0;
   for (0..$#x) {
      my ($x,$y) = (int($x[$_]*$f),$y[$_]);
      if ($x > $x0) {
         my $a = $pdl->slice("$x0:".($x-1));
         if ($y0 != $y) {
            $a .= $a->xlinvals($y0,$y);
         } else {
            $a .= $y;
         }
      }
      ($x0,$y0)=($x,$y);
   }
   $pdl;
}

=head2 gen_adsr duration*, sustain-level, attack-time, decay-time, sustain-time, release-time

Simple ADSR envelope generator. The C<sustain-level> is the amplitude (0
to 1) of the sustain level. The other for parameters give the relative
interval times, in any unit you like, only their relative ratios
are important. Any of these times might be zero, in which case the
corresponding part is omitted from the envelope.

=cut

sub gen_adsr($$$$$$) {
   my ($dur, $amp, $a, $d, $s, $r) = @_;
   my (@x, @y);
   my $t = 0;
   push @x, 0; push @y, 0;
   if ($a > 0) { push @x, $t += $a; push @y,    1 } else { $y[-1] =    1 }
   if ($d > 0) { push @x, $t += $d; push @y, $amp } else { $y[-1] = $amp }
   if ($s > 0) { push @x, $t += $s; push @y, $amp } else { $y[-1] = $amp }
   if ($r > 0) { push @x, $t += $r; push @y,    0 } else { $y[-1] =    0 }
   zeroes($dur)->xlinvals(0,$t)->linear_interpolate(pdl(@x), pdl(@y));
}

=head2 gen_asymmetric_fm duration*, freq/, phase, [r , [ratio]]

C<gen_asymmetric_fm> provides a way around the symmetric spectra normally
produced by FM. See Palamin and Palamin, "A Method of Generating and
Controlling Asymmetrical Spectra" JAES vol 36, no 9, Sept 88, p671-685.

=cut

sub gen_asymmetric_fm($$;$$$) {
   my ($dur, $freq, $phase, $r, $ratio) = @_;
   $r     ||= 1;
   $ratio ||= 1;
   my $cosr = 0.5 * ($r - 1/$r);
   my $sinr = 0.5 * ($r + 1/$r);
   $phase = zeroes($dur)->xvals*$freq + $phase;
   my $mth = $phase * $ratio;
   exp ($cosr * cos $mth) * sin ($phase + $sinr * sin $mth);
}

=head2 gen_sum_of_cosines duration*, freq/, phase, ncosines, [fm_mod/]

Generates a sum of C<n> cosines C<(1 + 2(cos(x) + cos(2x) + ... cos(nx)) =
sin((n+.5)x) / sin(x/2))>. Other arguments are similar to to C<gen_oscil>.

=head2 gen_sine_summation duration*, freq/, phase, [nsines, [a, [b_ratio, [fm_mod/]]]]

C<gen_sine_summation> provides a kind of additive synthesis. See
J.A.Moorer, "Signal Processing Aspects of Computer Music" and "The
Synthesis of Complex Audio Spectra by means of Discrete Summation
Formulae" (Stan-M-5). The basic idea is very similar to that used in
gen_sum_of_cosines generator.

The default value for C<nsines> is 1 (but zero is a valid value), for C<a>
is 0.5 and for C<b_ratio> is 1.

(btw, either my formula is broken or the output indeed does not lie
between -1 and +1, but rather -5 .. +5).

=cut

sub gen_sum_of_cosines {
   my ($dur, $freq, $phase, $cosines, $fm_mod) = @_;
   $cosines ||= 1;
   $dur = M_2PI*_dur2time($dur,$freq,$phase+1e-6,$fm_mod);
   sin ($dur * ($cosines+0.5)) / (sin ($dur * 0.5) * (1+2*$cosines));
}

sub gen_sine_summation {
   my ($dur, $freq, $phase, $n, $a, $ratio, $fm_mod) = @_;
   $n = 1 unless defined $n;
   $a ||= 0.5;
   $ratio ||= 1;

   $dur = M_2PI*_dur2time($dur,$freq,$phase+1e-6,$fm_mod);
   my $an = $n ? $a**($n+1) : 0;
   my $B  = $dur * $ratio;
   (sin ($dur)
    - $a * sin ($dur - $B)
    - $an * (sin ($dur + $B*($n+1))
             - $a * sin ($dur + $B*$n)))
   / (1 + $a*$a - (2*$a*cos($B)));
}

=head2 gen_from_table duration*, frequency/, table, [phase], [fm_mod/]

C<gen_from_table> generates a waveform by repeating a waveform given
in C<table>, linearly interpolating between successive points of the
C<waveform>.

=head2 partials2waveshape size*, partials, amplitudes, [phase], [fm_mod/]

Take a (perl or pdl) list of (integer) C<partials> and a list of
C<amplitudes> and generate a single wave shape that results by adding
these partial sines.

This could (and should) be used by the C<gen_from_table> generator.

=head2 gen_from_partials duration*, frequency/, partials, amplitudes, [phase], [fm_mod/]

Take a (perl list or pdl) list of (possibly noninteger) C<partials> and a
list of C<amplitudes> and generate the waveform resulting by summing up
all these partial sines.

=cut

sub gen_from_table {
   my ($dur, $freq, $table, $phase, $fm_mod) = @_;
   my $r = _dur2time($dur,$freq,$phase,$fm_mod);
   my $tx = ($table->dims)[0];
   $table = cat($table, $table->at(0))->clump(-1);
   $r %= 1; $r += 1; $r %= 1; $r *= $tx;
   $r->linear_interpolate($table->xvals, $table);
}

sub partials2waveshape {
   my $dur = zeroes $_[0];
   my @idx = ref $_[1] eq "ARRAY" ? @{$_[1]} : $_[1]->list;
   my @amp = ref $_[2] eq "ARRAY" ? @{$_[2]} : $_[2]->list;

   my $angle = $dur->xvals * (M_2PI/$dur->getdim(0));

   for (0..$#idx) {
      $dur += $amp[$_] * sin $idx[$_]*$angle;
   }
   $dur / max abs $dur;
}

sub gen_from_partials {
   my ($dur, $freq, $ti, $ta, $phase, $fm_mod) = @_;
   my @idx = ref $ti eq "ARRAY" ? @{$ti} : $ti->list;
   my @amp = ref $ta eq "ARRAY" ? @{$ta} : $ta->list;

   $dur = zeroes $dur;

   my $angle = _dur2time($dur,$freq,$phase,$fm_mod);

   for (0..$#idx) {
      $dur += $amp[$_] * fast_sin($idx[$_]*$angle);
   }
   $dur / max abs $dur;
}

EOP

pp_def '_gen_noise_1f',
       Pars => '[o]f(n)',
       GenericTypes => [F,D],
       Doc => undef,
       PMCode => undef,
       Code => q^
         static unsigned long ctz[64] = {
           6, 0, 1, 0, 2, 0, 1, 0,
           3, 0, 1, 0, 2, 0, 1, 0,
           4, 0, 1, 0, 2, 0, 1, 0,
           3, 0, 1, 0, 2, 0, 1, 0,
           5, 0, 1, 0, 2, 0, 1, 0,
           3, 0, 1, 0, 2, 0, 1, 0,
           4, 0, 1, 0, 2, 0, 1, 0,
           3, 0, 1, 0, 2, 0, 1, 0,
         };
         static unsigned int dice[7];
         static unsigned int total = 0;
         unsigned int n, prevrand, newrand, seed, k; 

         threadloop %{
           for (n = 0; n < $SIZE(n); n++)
             {
               k = ctz [n & 63];

               prevrand = dice[k];
               seed = 1664525 * seed + 1013904223;
               newrand = (seed >> 3);
               dice[k] = newrand;

               total += newrand - prevrand;

               seed = 1103515245 * seed + 12345;
               newrand = (seed >> 3);

               $f(n=>n) = ($GENERIC())(total + newrand) * (1./(3<<29)) - 1;
             }
         %}
         ^
;

pp_def 'filter_one_zero',
       Pars => 'in(n); [o] out(n)',
       OtherPars => 'double a0; double a1',
       GenericTypes => [F,D],
       Doc => 'apply a one zero filter, y(n) = a0 x(n) + a1 x(n-1)',
       Code => q!
          double a0 = $COMP(a0);
          double a1 = $COMP(a1);
          double x1 = 0.;

          threadloop %{ loop(n) %{
             $out() = a0 * $in() + a1 * x1;
             x1 = $in();
          %} %}
       !
;

pp_def 'filter_one_pole',
       Pars => 'in(n); [o] out(n)',
       OtherPars => 'double a0; double b1',
       GenericTypes => [F,D],
       Doc => 'apply a one pole filter, y(n) = a0 x(n) - b1 y(n-1)',
       Code => q!
          double a0 = $COMP(a0);
          double b1 = $COMP(b1);
          double y2 = 0.;

          threadloop %{ loop(n) %{
             $out() = y2 = a0 * $in() - b1 * y2;
          %} %}
       !
;

pp_def 'filter_two_zero',
       Pars => 'in(n); [o] out(n)',
       OtherPars => 'double a0; double a1; double a2',
       GenericTypes => [F,D],
       Doc => 'apply a two zero filter, y(n) = a0 x(n) + a1 x(n-1) + a2 x(n-2)',
       Code => q!
          double a0 = $COMP(a0);
          double a1 = $COMP(a1);
          double a2 = $COMP(a2);
          double x1 = 0.;
          double x2 = 0.;

          threadloop %{ loop(n) %{
             $out() = a0 * $in() + a1 * x1 + a2 * x2;
             x2 = x1; x1 = $in();
          %} %}
       !
;

pp_def 'filter_two_pole',
       Pars => 'in(n); [o] out(n)',
       OtherPars => 'double a0; double b1; double b2',
       GenericTypes => [F,D],
       Doc => 'apply a two pole filter, y(n) = a0 x(n) - b1 y(n-1) - b2 y(n-2)',
       Code => q!
          double a0 = $COMP(a0);
          double b1 = $COMP(b1);
          double b2 = $COMP(b2);
          double y1 = 0.;
          double y2 = 0.;

          threadloop %{ loop(n) %{
             $out() = a0 * $in() - b1 * y1 - b2 * y2;
             y2 = y1; y1 = $out();
          %} %}
       !
;

pp_def 'filter_formant',
       Pars => 'in(n); [o] out(n)',
       OtherPars => 'double radius; double frequency; double gain',
       GenericTypes => [F,D],
       Doc => 'apply a formant filter, y(n) = x(n) - r*x(n-2) + 2*r*cos(2*pi*frequency)*y(n-1) - r*r*y(n-2). A good value for C<gain> is 1.',
       Code => q!
          threadloop %{
             double radius = 1. - $COMP(radius);
             double a0 = $COMP(gain) * sin ($COMP(frequency) * M_2PI) * (1. - (radius * radius));
             double a2 = -radius;
             double b1 = -2. * radius * cos ($COMP(frequency) * M_2PI);
             double b2 = radius * radius;
             double x1 = 0.;
             double x2 = 0.;
             double y1 = 0.;
             double y2 = 0.;

             loop(n) %{
                double inval = a0 * $in();
                $out() = inval + a2 * x2 - b1 * y1 - b2 * y2;
                y2 = y1; y1 = $out();
                x2 = x1; x1 = inval;
             %}
          %}
       !
;

pp_addpm <<'EOP';

=head2 filter_ppolar pdl, radius/, frequency/

apply a two pole filter (given in polar form). The filter has two poles,
one at (radius,frequency), the other at (radius,-frequency). Radius
is between 0 and 1 (but less than 1), and frequency is between 0 and
0.5. This is the standard resonator form with poles specified by the
polar coordinates of one pole.

=head2 filter_zpolar pdl, radius/, frequency/

apply a two zero filter (given in polar form). See C<filter_ppolar>.

=cut

sub filter_ppolar {
   my ($pdl, $radius, $freq) = @_;
   $pdl->filter_two_pole(1, -2*$radius*cos($freq*M_2PI), $radius**2);
}

sub filter_zpolar {
   my ($pdl, $radius, $freq) = @_;
   $pdl->filter_two_zero(1, -2*$radius*cos($freq*M_2PI), $radius**2);
}

=head2 partials2polynomial partials, [kind]

C<partials2polynomial> takes a list of harmonic amplitudes and returns a
list of Chebychev polynomial coefficients. The argument C<kind> determines
which kind of Chebychev polynomial we are interested in, 1st kind or 2nd
kind. (default is 1).

=cut

sub partials2polynomial {
   my ($partials, $kind) = @_;
   $kind = 1 unless defined $kind;
   my $t0  = zeroes($partials->getdim(0)+1);
   my $t1  = $t0->copy;
   my $cc1 = $t0->copy;

   $t0->set(0,1);
   $t1->set(1,$kind);

   for my $amp ($partials->list) {
      $cc1 += $amp * $t1;
      $tn = 2 * $t1->rshift(1) - $t0;
      $t0 = $t1;
      $t1 = $tn;
   }
   $cc1;
}

=head2 ring_modulate in1, in2

ring modulates in1 with in2 (this is just a multiply).

=head2 amplitude_modulate am_carrier, in1, in2

amplitude modulates am_carrier and in2 with in1 (this calculates in1 * (am_carrier + in2)).

=cut

sub ring_modulate {
   $_[0] * $_[1];
}

sub amplitude_modulate {
   $_[1] * ($_[0] + $_[2]);
}

EOP

pp_def 'filter_sir',
       Pars => 'x(n); a(an); b(bn); [o]y(n)',
       GenericTypes => [L,F,D],
       Doc => <<'EOD',
Generic (short delay) impulse response filter. C<x> is the input signal
(which is supposed to be zero for negative indices). C<a> contains the
input (x) coefficients (a0, a1, .. an), whereas C<b> contains the output
(y) coefficients (b0, b1, ... bn), i.e.:

 y(n) = a0 x(n) - b1 y(n-1) + a1 x(n-1) - b2 y(n-2) + a2 x(n-2) - b3 ...

This can be used to generate fir and iir filters of any length, or even
more complicated constructs.

C<b0> (then first element of C<b>) is being ignored currently AND SHOULD
BE SPECIFIED AS ONE FOR FUTURE COMPATIBILITY
EOD
       Code => q^
          int an = $SIZE(an);
          int bn = $SIZE(bn);
          int i, n, I;
          $GENERIC() v;

          threadloop %{
            for (n = 0; n < $SIZE(n); n++)
              {
                v = 0;

                /* apply loop splitting manually later! */

                for (i = 0; i < an; i++)
                  {
                    I = n-i;
                    if (I >= 0)
                      v += $a(an=>i) * $x(n=>I);
                  }

                for (i = 0; i < bn; i++)
                  {
                    I = n-i;
                    if (I >= 0)
                      v -= $b(bn=>i) * $y(n=>I);
                  }

                $y(n=>n) = v;
              }
          %}
       ^
;

pp_def 'filter_lir',
       Pars => 'x(n); int a_x(an); a_y(an); int b_x(bn); b_y(bn); [o]y(n)',
       GenericTypes => [L,F,D],
       Doc => <<'EOD',

Generic (long delay) impulse response filter. The difference to
C<filter_sir> is that the filter coefficients need not be consecutive,
but instead their indices are given by the C<a_x> and C<b_x> (integer)
vectors, while the corresponding coefficients are in C<a_y> and
C<b_y>. (All C<a_x> must be >= 0, while all the C<b_x> must be >= 1, as
you should expect).

See C<filter_sir> for more info.
EOD
       Code => q^
          int i, n, I;
          $GENERIC() v;

          threadloop %{
            for (n = 0; n < $SIZE(n); n++)
              {
                v = 0;

                loop(an) %{
                  I = n-$a_x();
                  if (I >= 0)
                    v += $a_y() * $x(n=>I);
                %}

                loop(bn) %{
                  I = n-$b_x();
                  if (I >= 0)
                    v -= $b_y() * $y(n=>I);
                %}

                $y(n=>n) = v;
              }
          %}
       ^
;

pp_addpm <<'EOP';

=head2 filter_fir input, xcoeffs

Apply a fir (finite impulse response) filter to C<input>. This is the same as
calling:

 filter_sir input, xcoeffs, pdl()

=head2 filter_iir input, ycoeffs

Apply a iir (infinite impulse response) filter to C<input>. This is just
another way of saying:

 filter_sir input, pdl(1), ycoeffs

That is, the first member of C<ycoeffs> is being ignored AND SHOULD BE
SPECIFIED AS ONE FOR FUTURE COMPATIBILITY!

=cut

sub filter_fir($$) {
   filter_sir ($_[0], $_[1], PDL->pdl());
}

sub filter_iir($$) {
   filter_sir ($_[0], PDL->pdl(1), $_[1]);
}

=head2 filter_comb input, delay*, scaler

Apply a comb filter to the piddle C<input>. This is implemented using a
delay line of length C<delay> (which must be 1 or larger and can be
non-integer) and a feedback scaler.

 y(n) = x(n-size-1) + scaler * y(n-size)

cf. C<filter_notch> and http://www.harmony-central.com/Effects/Articles/Reverb/comb.html

=head2 filter_notch input, delay*, scaler

Apply a comb filter to the piddle C<input>. This is implemented using a
delay line of length C<delay> (which must be 1 or larger and can be
non-integer) and a feedforward scaler.

 y(n) = x(n-size-1) * scaler + y(n-size)

As a rule of thumb, the decay time of the feedback part is
C<7*delay/(1-scaler)> samples, so to get a decay of Dur seconds,
C<scaler <= 1-7*delay/(Dur*Srate)>. The peak gain is C<1/(1-(abs
scaler))>. The peaks (or valleys in notch's case) are evenly spaced at
C<srate/delay>. The height (or depth) thereof is determined by scaler
-- the closer to 1.0, the more pronounced. See Julius Smith's "An
Introduction to Digital Filter Theory" in Strawn "Digital Audio Signal
Processing", or Smith's "Music Applications of Digital Waveguides"

=head2 filter_allpass input, delay*, scaler-feedback, scaler-feedforward

C<filter_allpass> or "moving average comb" is just like C<filter_comb>
but with an added feedforward term. If C<scaler-feedback == 0>, we get a
moving average comb filter. If both scaler terms == 0, we get a pure delay
line.

 y(n) = feedforward*x(n-1) + x(n-size-1) + feedback*y(n-size)

cf. http://www.harmony-central.com/Effects/Articles/Reverb/allpass.html

=cut

sub _filter_combnotchall {
   my ($pdl, $size, $scalercomb, $scalernotch, $scalerall) = @_;
   my $D = int $size;
   my $F = $size - $D;
   my $a = zeroes $D+2; $a->set($D,1-$F); $a->set($D+1, $F);
   my $b = zeroes $D+2; $b->set($D,$F-1); $b->set($D+1,-$F);
   $a->set(1, $a->at(1) + $scalerall);
   $b->set(0, 1);
   $pdl->filter_sir($a*$scalernotch,$b*$scalercomb);
}

sub filter_comb {
   _filter_combnotchall $_[0], $_[1], $_[2], 1, 0;
}

sub filter_notch {
   _filter_combnotchall $_[0], $_[1], 1, $_[2], 0;
}

sub filter_allpass {
   _filter_combnotchall $_[0], $_[1], $_[2], 1, $_[3];
}

=head2 design_remez_fir filter_size, bands(2,b), desired_gain(b), type, [weight(b)]
       
Calculates the optimal (in the Chebyshev/minimax sense) FIR filter
impulse response given a set of band edges, the desired reponse on those
bands, and the weight given to the error in those bands, using the
Parks-McClellan exchange algorithm.

The first argument sets the filter size: C<design_remez_fir> returns as
many coefficients as specified by this parameter.

C<bands> is a vector of band edge pairs (start - end), which specify the
start and end of the bands in the filter specification.  These must be
non-overlapping and sorted in increasing order. Only values between C<0>
(0 Hz) and C<0.5> (the Nyquist frequency) are allowed.

C<des> specifies the desired gain in these bands.

C<weight> can be used to give each band a different weight. If absent, a
vector of ones is used.

C<type> is any of the exported constants C<BANDPASS>, C<DIFFERENTIATOR> or
C<HILBERT> and can be used to select various design types (use C<BANDPASS>
until this is documented ;)

=cut

sub design_remez_fir {
   my ($size, $bands, $des, $type, $weight) = @_;
   $weight = ones $des unless defined $weight;
   $size = zeroes $size;
   _design_remez_fir($bands, $des, $weight, $size, $type);
   $size;
}

=head2 filter_src input, srate, [width], [sr-mod]

Generic sampling rate conversion, implemented by convoluting C<input> with
a sinc function of size C<width> (default when unspecified or zero: 5).

C<srate> determines the input rate / output rate ratio, i.e. values > 1
speed up, values < 1 slow down. Values < 0 are allowed and reverse the
signal.

If C<sr_mod> is omitted, the size of the output piddle is calculcated
as C<length(input)/abs(srate)>, e.g. it provides the full stretched or
shrinked input signal.

If C<sr_mod> is specified it must be as large as the desired output, i.e.
it's size determines the output size. Each value in C<sr_mod> is added to
C<srate> at the given point in "time", so it can be used to "modulate" the
sampling rate change.

 # create a sound effect in the style of "Forbidden Planet"
 $osc = 0.3 * gen_oscil $osc, 30 / $pdl->rate;
 $output = filter_src($input, 1, 0, $osc);

=cut

sub filter_src($$;$$) {
   my ($input, $srate, $width, $sr_mod) = @_;

   $width ||= 5;
   $width*2 < $input->getdim(0) or barf "src width too large (> 0.5 * input size)";

   my $output;

   if (defined $sr_mod) {
      $output = zeroes $sr_mod;
   } else {
      $output = zeroes ($input->getdim(0) / abs($srate));
      $sr_mod = PDL->null;
   }

   _filter_src($input, $output, $sr_mod, $srate, $width);

   $output->rate($input->rate / abs($srate)) if $input->rate;
   $output;
}

=head2 filter_contrast_enhance input, enhancement

Contrast-enhancement phase-modulates a sound file. It's like audio
MSG. The actual algorithm is (applied to the normalised sound)
C<sin(input*pi/2 + (enhancement*sin(input*2*pi)))>. The result is to
brighten the sound, helping it cut through a huge mix.

=cut

sub filter_contrast_enhance {
   my ($pdl, $idx) = @_;
   $idx = 0.1 unless defined $idx;
   $pdl = $pdl - min $pdl;
   $pdl = ($pdl * (M_PI / max $pdl)) - M_PI * 0.5;
   sin $pdl + $idx * sin $pdl*4;
}

=head2 filter_granulate input, expansion, [option-hash], option => value...

C<filter_granulate> "granulates" the sound file file. It is the poor
man's way to change the speed at which things happen in a recorded sound
without changing the pitches. It works by slicing the input file into
short pieces, then overlapping these slices to lengthen (or shorten) the
result; this process is sometimes known as granular synthesis, and is
similar to the "freeze" function. The duration of each slice is C<length> --
the longer, the more like reverb the effect. The portion of the length (on
a scale from 0 to 1.0) spent on each ramp (up or down) is C<ramp>. This can
control the smoothness of the result of the overlaps. The more-or-less
average time between successive segments is C<hop>.  The accuracy at which we
handle this hopping is set by the float C<jitter> -- if C<jitter> is very small,
you may get an annoying tremolo. The overall amplitude scaler on each
segment is C<scaler> -- this is used to try to to avoid overflows as we add
all these zillions of segments together. C<expansion> determines the input
hop in relation to the output hop; an expansion-amount of C<2.0> should more
or less double the length of the original, whereas an expansion-amount
of C<1.0> should return something close to the original speed.

The defaults for the arguments/options are:

 expansion      1.0
 length(*)      0.15
 scaler         0.6
 hop(*)         0.05
 ramp           0.4
 jitter(*)      0.5
 maxsize        infinity

The parameters/options marked with (*) actually depend on the sampling
rate, and are always multiplied by the C<rate> attribute of the
piddle internally. If the piddle lacks that attribute, 44100 is
assumed. NOTE: This is different to most other filters, but should be ok
since C<filter_granulate> only makes sense for audiofiles.

=cut

sub filter_granulate  {
   my $input = shift;
   my $expansion = shift || 1;
   my %hdr = (); %hdr = (%hdr, %{+shift}) if ref $_[0]; %hdr = (%hdr, @_) if @_;

   my $rate    =  $hdr{rate}   || 44100;
   my $length  = ($hdr{length} || 0.15) * $rate;
   my $scaler  =  $hdr{scaler} || 0.6;
   my $hop     = ($hdr{hop}    || 0.05) * $rate;
   my $ramp    =  $hdr{ramp}   || 0.4;
   my $jitter  = ($hdr{jitter} || 0.5) * $rate;
   my $maxsize =  $hdr{maxsize}|| 0;

   $output = zeroes $expansion * $input->getdim(0);

   _filter_granulate($input, $output,
                     $expansion, $length, $scaler,
                     $hop, $ramp, $jitter, $maxsize);
   $output;
}

EOP

pp_def '_design_remez_fir',
       Pars => 'bands(2,b); des(b); weight(b); [o]h(n)',
       OtherPars => 'int type',
       GenericTypes => [D],
       Doc => undef,
       PMCode => undef,
       Code => q^
          int type = $COMP(type);

          if (type != BANDPASS
              && type != DIFFERENTIATOR
              && type != HILBERT)
            barf ("design_remez_fir: illegal type specified (none of BANDPASS, DIFFERENTIATOR, HILBERT)");

          remez ($P(h), $SIZE(n), $SIZE(b), $P(bands), $P(des), $P(weight), type);
       ^
;

pp_def '_filter_src',
       Pars => 'input(n); [o]output(m); sr_mod(m2)',
       OtherPars => 'double srate; int width',
       GenericTypes => [D],
       Doc => undef,
       PMCode => undef,
       Code => q!
          mus_src ($P(input), $SIZE(n), $P(output), $SIZE(m),
                   $COMP(srate),
                   $SIZE(m) == $SIZE(m2) ? $P(sr_mod) : 0,
                   $COMP(width));
       !
;

pp_def '_filter_granulate',
       Pars => 'input(n); [o]output(m)',
       OtherPars => 'double expansion; double length; double scaler; double hop; double ramp; double jitter; int max_size',
       GenericTypes => [D],
       Doc => undef,
       PMCode => undef,
       Code => q!
          mus_granulate ($P(input), $SIZE(n), $P(output), $SIZE(m),
                         $COMP(expansion), $COMP(length), $COMP(scaler),
                         $COMP(hop), $COMP(ramp), $COMP(jitter),
                         $COMP(max_size));
       !
;

pp_addpm <<'EOP';

=head2 audiomix pos1, data1, pos2, data2, ...

Generate a mix of all given piddles. The resulting piddle will contain the
sum of all data-piddles at their respective positions, so some scaling
will be necessary before or after the mixing operation (e.g. scale2short).

 # mix the sound gong1 at position 0, the sound bass5 at position 22100
 # and gong2 at position 44100. The resulting piddle will be large enough
 # to accomodate all the sounds:
 $mix = audiomix 0, $gong1, 44100, $gong2, 22100,  $gong2

=cut

sub audiomix {
   my(@x,@p);
   my($dur,$i,$end,$s,$p);
   for ($i = 0; $i < @_; $i += 2) {
      $end = $_[$i] + $_[$i+1]->getdim(0);
      $dur = $end if $end > $dur;
   }
   my $pdl = zeroes $dur;
   for ($i = 0; $i < @_; $i += 2) {
      ($s,$p) = (int($_[$i]), $_[$i+1]);
      ($end = $pdl->slice("$s:".($s+$p->getdim(0)-1))) += $p;
   }
   $pdl;
}

=head2 filter_center piddle

Normalize the piddle so that it is centered around C<y = 0> and has
maximal amplitude of 1.

=cut

sub filter_center($) {
   my $piddle = shift;
   $piddle = $piddle - min $piddle;
   $piddle * (2 / max $piddle) - 1;
}

=head2 scale2short piddle

This method takes a sound in any format (preferably float or double) and
scales it to fit into a signed short value, suitable for playback using
C<playudio> or similar functions.

=cut

sub scale2short {
   my $pdl = shift->float;
   ($pdl * (1 / max abs $pdl) * 32767.5 - 0.5)->short;
}

=head2 gen_fft_window size*, type, [$beta]

Creates and returns a specific fft window. The C<type> is any of the following.
These are (case-insensitive) strings, so you might need to quote them.

 RECTANGULAR    just ones (the identity window)
 HANNING        0.50 - 0.50 * cos (0 .. 2pi)
 HAMMING        0.54 - 0.46 * cos (0 .. 2pi)
 WELCH          1 - (-1 .. 1) ** 2
 PARZEN         the triangle window
 BARTLETT       the symmetric triangle window
 BLACKMAN2      blackman-harris window of order 2
 BLACKMAN3      blackman-harris window of order 3
 BLACKMAN4      blackman-harris window of order 4
 EXPONENTIAL    the exponential window
 KAISER         the kaiser/bessel window (using the parameter C<beta>)
 CAUCHY         the cauchy window (using the parameter <beta>)
 POISSON        the poisson window (exponential using parameter C<beta>)
 RIEMANN        the riemann window (sinc)
 GAUSSIAN       the gaussian window of order C<beta>)
 TUKEY          the tukey window (C<beta> specifies how much of the window
                consists of ones).
 COSCOS         the cosine-squared window (a partition of unity)
 SINC           same as RIEMANN
 HANN           same as HANNING (his name was Hann, not Hanning)

 LIST           this "type" is special in that it returns a list of all types

=cut

sub gen_fft_window {
   my ($size, $type, $beta) = @_;

   $beta = 2.5 unless defined $beta;

   $size = $size->getdim(0) if ref $size;
   $size > 2 or barf "fft window size too small";

   my $midn  = $size >> 1;
   my $midm1 = ($size-1) >> 1;
   my $midp1 = ($size+1) >> 1;
   my $dur   = zeroes $size;
   my $sf    = ($size-1)/$size;
   %fft_window = (
      RECTANGULAR => sub {
         $dur->ones
      },
      HANNING => sub {
         0.5 - 0.5 * cos $dur->xlinvals(0,M_2PI*$sf)
      },
      HAMMING => sub {
         0.53836 - 0.46164 * cos $dur->xlinvals(0,M_2PI*$sf)
      },
      WELCH => sub {
         1 - $dur->xlinvals(-1,$sf)**2;
      },
      PARZEN => sub {
         1 - abs $dur->xlinvals(-$sf,$sf);
      },
      BARTLETT => sub {
         1 - abs $dur->xlinvals(-1,1);
      },
      BLACKMAN2 => sub {
         my $cx = cos $dur->xlinvals(0,M_2PI*$sf);
         0.34401 + ($cx * (-0.49755 + ($cx * 0.15844)));
      },
      BLACKMAN3 => sub {
         my $cx = cos $dur->xlinvals(0,M_2PI*$sf);
         0.21747 + ($cx * (-0.45325 + ($cx * (0.28256 - ($cx * 0.04672)))));
      },
      BLACKMAN4 => sub {
         my $cx = cos $dur->xlinvals(0,M_2PI*$sf);
         0.084037 + ($cx * (-0.29145 + ($cx * (0.375696 + ($cx * (-0.20762 + ($cx * 0.041194)))))));
      },
      EXPONENTIAL => sub {
         2 ** (1 - abs $dur->xlinvals(-1,$sf)) - 1;
      },
      KAISER => sub {
         bessi0 ($beta * sqrt(1 - $dur->xlinvals(-1,$sf)**2)) / bessi0 ($beta);
      },
      CAUCHY => sub {
         1 / (1 + ($dur->xlinvals(-1,$sf)*$beta)**2);
      },
      POISSON => sub {
         exp (-$beta * abs $dur->xlinvals(-1,$sf));
      },
      RIEMANN => sub {
         my $dur1 = $dur->slice("0:$midm1");
         my $dur2 = $dur->slice("-1:$midp1:-1");
         my $cx;
         $cx = $dur1->xlinvals(M_PI,M_2PI/$size); $dur1 .= sin ($cx) / $cx;
         $cx = $dur2->xlinvals(M_PI,M_2PI/$size); $dur2 .= sin ($cx) / $cx;
         ($cx = $dur->slice("$midm1:$midp1")) .= 1;
         $dur;
      },
      GAUSSIAN => sub {
         exp (-0.5 * ($beta * abs $dur->xlinvals(-1,$sf))**2);
      },
      TUKEY => sub {
         $beta >= 0 && $beta <= 1 or barf "beta must be between 0 and 1 for the tukey window";
         my $cx = int ($midn * (1 - $beta));
         my $cX = $size-$cx-1;
         my $dur1 = $dur->slice("0:".($cx-1));
         my $dur2 = $dur->slice("-1:".($cX+1).":-1");
         my $dur3 = $dur->slice("$cx:$cX");

         $dur1 .= 0.5 * (1 - cos $dur1->xlinvals(0, M_PI));
         $dur2 .= 0.5 * (1 - cos $dur2->xlinvals(0, M_PI));
         $dur3 .= 1;

         $dur;
      },
      COSCOS => sub {
         (cos $dur->xlinvals(M_PI/-2,M_PI/2))**2;
      },
      LIST => sub {
         grep $_ ne "LIST", keys %fft_window;
      },
   );

   $fft_window{SINC} = $fft_window{RIEMANN};
   $fft_window{HANN} = $fft_window{HANNING};

   $type = uc $type;
   $fft_window{$type} or barf "$type: no such window";
   $fft_window{$type}->();
}

=head2 cplx(2,n) = rfft real(n)

Do a (complex fft) of C<real> (extended to complex so that the imaginary
part is zero), and return the complex fft result. This function tries
to use L<PDL::FFTW|PDL::FFTW> (which is faster for large vectors) when
available, and falls back to L<PDL::FFT|PDL::FFT>, which is likely
to return different phase signs (due to different kernel functions),
so beware! In fact, since C<rfft> has to shuffle the data when using
PDL::FFT, the fallback is always slower.

When using PDL::FFTW, a wisdom file ~/.pdl_wisdom is used and updated, if
possible.

=head2 real(n) = irfft cplx(2,n)

The inverse transformation (see C<rfft>). C<irfft rfft $pdl == $pdl>
always holds.

=cut

my $fftw_loaded;
sub _fftw {
   defined $fftw_loaded or eval {
      $fftw_loaded = 0;
      require PDL::FFTW;
      PDL::FFTW::load_wisdom("$ENV{HOME}/.pdl_wisdom")
         if -r "$ENV{HOME}/.pdl_wisdom";
      $fftw_loaded = 1;
   };
   $fftw_loaded;
}

sub rfft {
   my $data = $_[0];
   if (_fftw) {
      my $x = $data->r2C;
      $x = PDL::FFTW::fftw $x;
      $x;
   } else {
      require PDL::FFT;
      my @fft = ($data->copy, $data->zeroes);
      PDL::FFT::fft(@fft);
      cat(@fft)->xchg(0,1);
   }
}

sub irfft {
   if (_fftw) {
      $x = $_[0]->copy;
      $x = PDL::FFTW::ifftw $x;
      re $x / $x->getdim(1);
   } else {
      require PDL::FFT;
      my @fft = $_[0]->xchg(0,1)->dog({Break => 1});
      PDL::FFT::ifft(@fft);
      $fft[0];
   }
}

=head2 spectrum data, [norm], [window], [beta]

Returns the spectrum of a given pdl. If C<norm> is absent (or C<undef>),
it returns the magnitude of the fft of C<data>. When C<norm> == 1 (or
C<eq 'NORM'>, case-insensitive), it returns the magnitude, normalized to be
between zero and one. If C<norm> == 0 (or C<eq 'dB'>, case-insensitive), then
it returns the magnitude in dB.

C<data> is multiplied with C<window> (if not C<undef>) before calculating
the fft, and usually contains a window created with C<gen_fft_window>
(using C<beta>). If C<window> is a string, it is handed over to
C<gen_fft_window> (together with the beta parameter) to create a window of
suitable size.

This function could be slightly faster.

=cut

sub spectrum {
   my ($data, $norm, $window,  $beta) = @_;
   my $len;
   if (defined $window) {
      $window = gen_fft_window ($data->getdim (0), $window, $beta) unless ref $window;
      $data = $data * $window;
      $len = $window->getdim (0);
   } else {
      $len = $data->getdim (0);
   }
   $data = rfft (
              $data->slice ("0:" . ($len - 1))
                   ->sever
           )
           ->slice (",0:" . int ($len / 2))
           ->PDL::Complex::Cr2p
           ->slice ("(0)");
   if ($norm == 1 || lc $norm eq "norm") {
      $data / max $data;
   } elsif (($norm =~ /^[.0]+$/) || (lc $norm eq "db")) {
      log (1e-37 + $data / max $data) * (20 / log 10);
   } else {
      $data;
   }
}

=head2 concat pdl, pdl...

This is not really an audio-related function. It simply takes all piddles
and concats them into a larger one. At the moment it only supports
single-dimensional piddles and is implemented quite slowly using perl and
data-copying, but that might change...

=cut

sub concat {
   my $len = sum pdl map $_->getdim(0), @_;
   my $r = zeroes $len;
   my $o = 0;
   for (@_) {
      my $e = $o + $_->getdim(0) - 1;
      (my $t = $r->slice("$o:$e")) .= $_;
      $o = $e + 1;
   }
   $r;
}

EOP

# TODO: document!
pp_def 'filter_convolve',
       Pars => 'input(n); kernel(m); int fftsize(); [o]output(n)',
       GenericTypes => [D],
       Code => q!
          mus_convolve ($P(input), $P(output), $SIZE(n), $P(kernel), $fftsize(), $SIZE(m));
       !
;

pp_def 'rshift',
        Doc => <<'EOD',
=for ref

Shift vector elements without wrap and fill the free space with a
constant. Flows data back & forth, for values that overlap.

Positive values shift right, negative values shift left.

EOD
        Pars => 'x(n); int shift(); c(); [oca]y(n)',
        DefaultFlow => 1,
        Reversible => 1,
        PMCode => '
           sub PDL::rshift {
              my @a = @_;
              if($#a == 3) {
                 &PDL::_rshift_int(@a);@a=();
              } elsif($#a == 1 || $#a == 2) {
                 $a[3] = PDL->nullcreate($a[0]);
                 &PDL::_rshift_int(@a);
                 $a[3];
              } else {
                 barf "Invalid number of arguments for shiftin";
              }
           }
        ',
        Code=>'
           int i,j;
           int n_size = $SIZE(n);
           for(i = -$shift(), j=0; j < n_size; i++, j++)
             $y(n=>j) = i >= 0 && i < n_size ? $x(n=>i) : $c();
        ',
        BackCode=>'
           int i,j;
           int n_size = $SIZE(n);
           for(i = -$shift(), j=0; j < n_size; i++, j++)
             if (i >= 0 && i < n_size)
                $x(n=>i) = $y(n=>j);
        '
;

pp_def 'polynomial',
       Pars => 'coeffs(n); x(m); [o]out(m)',
       Doc => 'evaluate the polynomial with coefficients C<coeffs> at the position(s) C<x>. C<coeffs[0]> is the constant term.',
       Code => q!
          loop(m) %{
             $GENERIC() x = 1;
             $GENERIC() o = 0;

             loop(n) %{
                o += $coeffs() * x;
                x *= $x();
             %}

             $out() = o;
          %}
       !
;

pp_def 'linear_interpolate',
       Pars => 'x(); fx(n); fy(n); [o]y()',
       GenericTypes => [L,F,D],
       Doc => '
Look up the ordinate C<x> in the function given by C<fx> and C<fy>
and return a linearly interpolated value (somewhat optimized for many
lookups).

C<fx> specifies the ordinates (x-coordinates) of the function and most be
sorted in increasing order. C<fy> are the y-coordinates of the function in
these points.
       ',
       Code => q!
          int d = 0;
          int D = $SIZE(n) - 1; /* a tribute to PP's stupidity */
          $GENERIC() xmin =$fx(n=>0);
          $GENERIC() xmax =$fx(n=>D);

          threadloop %{
             $GENERIC() x = $x();
             $GENERIC() x1, x2, y1, y2;

             if (x <= xmin)
               $y() = $fy(n=>0);
             else if (x >= xmax)
               $y() = $fy(n=>D);
             else
               {
                 while ($fx(n=>d) >  x) d--;
                 while ($fx(n=>d) <= x) d++;
                 /* 0       < d <= D     */
                 /* fx(d-1) < x <= fx(d) */
                 x2 = $fx(n=>d);
                 y2 = $fy(n=>d);
                 d--;
                 x1 = $fx(n=>d);
                 y1 = $fy(n=>d);
                 $y() = y1 + (x-x1)*(y2-y1)/(x2-x1);
               }
          %}
       !
;

pp_def 'bessi0',
       Pars => 'a(); [o]b()',
       Doc => 'calculate the (approximate) modified bessel function of the first kind',
       GenericTypes => [F,D],
       Code => q!
          $GENERIC() x = $a();
          if (x > -3.75 && x < 3.75)
            {
              $GENERIC() y = (x / 3.75); y *= y;
              $b() = 1 + (y * (3.5156229 +
                               (y * (3.0899424 +
                                       (y * (1.2067492 +
                                               (y * (0.2659732 +
                                                       (y * (0.360768e-1 +
                                                               (y * 0.45813e-2)))))))))));
            }
          else
            {
              $GENERIC() ax = x < 0 ? -x : x;
              $GENERIC() y  = ax / 3.75;
              $b() = ((exp(ax) / sqrt(ax)) *
                        (0.39894228 +
                           (y * (0.1328592e-1 +
                                   (y * (0.225319e-2 +
                                           (y * (-0.157565e-2 +
                                                   (y * (0.916281e-2 +
                                                           (y * (-0.2057706e-1 +
                                                                   (y * (0.2635537e-1 +
                                                                           (y * (-0.1647633e-1 +
                                                                                   (y * 0.392377e-2)))))))))))))))));
            }
       !,
;

pp_def 'fast_sin',
       Pars => 'r(n); [o]s(n)',
       GenericTypes => [F,D],
       Doc => 'fast sine function (inaccurate table lookup with ~12 bits precision)',
       Code => q^
#        define SINE_SIZE 16384
         static float *sine_table = 0;

         if (!sine_table)
           {
             int i;
             Float phase, incr;
             sine_table = (float *) calloc (SINE_SIZE + 1, sizeof (float));
             incr = M_2PI / (Float) SINE_SIZE;
             for (i = 0, phase = 0.0; i < SINE_SIZE + 1; i++, phase += incr)
               sine_table[i] = (float) sin (phase);
           }

         threadloop %{
           loop(n) %{
             $s() = sine_table[((int)($r() * (SINE_SIZE / M_2PI)) % SINE_SIZE + SINE_SIZE) % SINE_SIZE];
           %}
         %}^
;

pp_addpm {At => Bot}, <<'EOD';

=head1 AUTHOR

Marc Lehmann <schmorp@schmorp.de>. The ideas were mostly taken from common
lisp music (CLM), by Bill Schottstaedt C<bil@ccrma.stanford.edu>. I also
borrowed many explanations (and references) from the clm docs and some
code from clm.c. Highly inspiring!

=head1 SEE ALSO

perl(1), L<PDL>.

=cut
EOD

pp_addxs  "\nINCLUDE: xlib.xs\n";

pp_done;

Revision:	1.9
Committed:	Tue Nov 8 18:56:06 2005 UTC (18 years, 6 months ago) by root
Branch:	MAIN
CVS Tags:	rel-1_1
Changes since 1.8:	+8 -4 lines
Log Message:	* empty log message *