/// <summary>
/// SID clocking with audio sampling - delta clocking picking nearest sample
/// </summary>
/// <param name="delta_t"></param>
/// <param name="buf"></param>
/// <param name="n"></param>
/// <param name="interleave"></param>
/// <returns></returns>
protected int clock_fast(CycleCount delta_t, short[] buf, int n, int interleave)
{
int s = 0;
for (; ;)
{
int next_sample_offset = sample_offset + cycles_per_sample + (1 << (FIXP_SHIFT - 1));
int delta_t_sample = next_sample_offset >> FIXP_SHIFT;
if (delta_t_sample > delta_t.delta_t)
{
break;
}
if (s >= n)
{
return(s);
}
clock(delta_t_sample);
delta_t.delta_t -= delta_t_sample;
sample_offset = (next_sample_offset & FIXP_MASK) - (1 << (FIXP_SHIFT - 1));
buf[s++ *interleave] = (short)output();
}
clock(delta_t.delta_t);
sample_offset -= delta_t.delta_t << FIXP_SHIFT;
delta_t.delta_t = 0;
return(s);
}
/// <summary>
/// SID clocking with audio sampling - cycle based with linear sample interpolation.
///
/// Here the chip is clocked every cycle. This yields higher quality sound
/// since the samples are linearly interpolated, and since the external
/// filter attenuates frequencies above 16kHz, thus reducing sampling noise
/// </summary>
/// <param name="delta_t"></param>
/// <param name="buf"></param>
/// <param name="n"></param>
/// <param name="interleave"></param>
/// <returns></returns>
protected int clock_interpolate(CycleCount delta_t, short[] buf, int n, int interleave)
{
int s = 0;
int i;
for (; ;)
{
int next_sample_offset = sample_offset + cycles_per_sample;
int delta_t_sample = next_sample_offset >> FIXP_SHIFT;
if (delta_t_sample > delta_t.delta_t)
{
break;
}
if (s >= n)
{
return(s);
}
for (i = 0; i < delta_t_sample - 1; i++)
{
clock();
}
if (i < delta_t_sample)
{
sample_prev = (short)output();
clock();
}
delta_t.delta_t -= delta_t_sample;
sample_offset = next_sample_offset & FIXP_MASK;
short sample_now = (short)output();
buf[s++ *interleave] = (short)(sample_prev + (sample_offset * (sample_now - sample_prev) >> FIXP_SHIFT));
sample_prev = sample_now;
}
for (i = 0; i < delta_t.delta_t - 1; i++)
{
clock();
}
if (i < delta_t.delta_t)
{
sample_prev = (short)output();
clock();
}
sample_offset -= delta_t.delta_t << FIXP_SHIFT;
delta_t.delta_t = 0;
return(s);
}
/// <summary>
/// SID clocking with audio sampling
/// </summary>
/// <param name="delta_t"></param>
/// <param name="buf"></param>
/// <param name="n"></param>
/// <param name="interleave"></param>
/// <returns></returns>
public int clock(CycleCount delta_t, short[] buf, int n, int interleave)
{
switch (sampling)
{
default:
case SIDDefs.sampling_method.SAMPLE_FAST:
return(clock_fast(delta_t, buf, n, interleave));
case SIDDefs.sampling_method.SAMPLE_INTERPOLATE:
return(clock_interpolate(delta_t, buf, n, interleave));
case SIDDefs.sampling_method.SAMPLE_RESAMPLE_INTERPOLATE:
return(clock_resample_interpolate(delta_t, buf, n, interleave));
case SIDDefs.sampling_method.SAMPLE_RESAMPLE_FAST:
return(clock_resample_fast(delta_t, buf, n, interleave));
}
}
/// <summary>
/// SID clocking with audio sampling - cycle based with audio resampling
/// </summary>
/// <param name="delta_t"></param>
/// <param name="buf"></param>
/// <param name="n"></param>
/// <param name="interleave"></param>
/// <returns></returns>
protected int clock_resample_fast(CycleCount delta_t, short[] buf, int n, int interleave)
{
int s = 0;
for (; ;)
{
int next_sample_offset = sample_offset + cycles_per_sample;
int delta_t_sample = next_sample_offset >> FIXP_SHIFT;
if (delta_t_sample > delta_t.delta_t)
{
break;
}
if (s >= n)
{
return(s);
}
for (int i = 0; i < delta_t_sample; i++)
{
clock();
sample[sample_index] = sample[sample_index + RINGSIZE] = (short)output();
++sample_index;
sample_index &= 0x3fff;
}
delta_t.delta_t -= delta_t_sample;
sample_offset = next_sample_offset & FIXP_MASK;
int fir_offset = sample_offset * fir_RES >> FIXP_SHIFT;
int fir_start = (fir_offset * fir_N);
int sample_start = (sample_index - fir_N + RINGSIZE);
// Convolution with filter impulse response.
int v = 0;
for (int j = 0; j < fir_N; j++)
{
v += sample[sample_start + j] * fir[fir_start + j];
}
v >>= FIR_SHIFT;
// Saturated arithmetics to guard against 16 bit sample overflow.
int half = 1 << 15;
if (v >= half)
{
v = half - 1;
}
else if (v < -half)
{
v = -half;
}
buf[s++ *interleave] = (short)v;
}
for (int i = 0; i < delta_t.delta_t; i++)
{
clock();
sample[sample_index] = sample[sample_index + RINGSIZE] = (short)output();
++sample_index;
sample_index &= 0x3fff;
}
sample_offset -= delta_t.delta_t << FIXP_SHIFT;
delta_t.delta_t = 0;
return(s);
}
/// <summary>
/// SID clocking with audio sampling - cycle based with audio resampling.
///
/// This is the theoretically correct (and computationally intensive) audio
/// sample generation. The samples are generated by resampling to the
/// specified sampling frequency. The work rate is inversely proportional to
/// the percentage of the bandwidth allocated to the filter transition band.
///
/// This implementation is based on the paper "A Flexible Sampling-Rate
/// Conversion Method", by J. O. Smith and P. Gosset, or rather on the
/// expanded tutorial on the "Digital Audio Resampling Home Page":
/// http://www-ccrma.stanford.edu/~jos/resample/
///
/// By building shifted FIR tables with samples according to the sampling
/// frequency, this implementation dramatically reduces the computational
/// effort in the filter convolutions, without any loss of accuracy. The
/// filter convolutions are also vectorizable on current hardware.
///
/// Further possible optimizations are: * An equiripple filter design could
/// yield a lower filter order, see
/// http://www.mwrf.com/Articles/ArticleID/7229/7229.html * The Convolution
/// Theorem could be used to bring the complexity of convolution down from
/// O(n*n) to O(n*log(n)) using the Fast Fourier Transform, see
/// http://en.wikipedia.org/wiki/Convolution_theorem * Simply resampling in
/// two steps can also yield computational savings, since the transition band
/// will be wider in the first step and the required filter order is thus
/// lower in this step. Laurent Ganier has found the optimal intermediate
/// sampling frequency to be (via derivation of sum of two steps): 2 *
/// pass_freq + sqrt [ 2 * pass_freq * orig_sample_freq * (dest_sample_freq -
/// 2 * pass_freq) / dest_sample_freq ]
/// </summary>
/// <param name="delta_t"></param>
/// <param name="buf"></param>
/// <param name="n"></param>
/// <param name="interleave"></param>
/// <returns></returns>
protected int clock_resample_interpolate(CycleCount delta_t, short[] buf, int n, int interleave)
{
int s = 0;
for (; ;)
{
int next_sample_offset = sample_offset + cycles_per_sample;
int delta_t_sample = next_sample_offset >> FIXP_SHIFT;
if (delta_t_sample > delta_t.delta_t)
{
break;
}
if (s >= n)
{
return(s);
}
for (int i = 0; i < delta_t_sample; i++)
{
clock();
sample[sample_index] = sample[sample_index + RINGSIZE] = (short)output();
++sample_index;
sample_index &= 0x3fff;
}
delta_t.delta_t -= delta_t_sample;
sample_offset = next_sample_offset & FIXP_MASK;
int fir_offset = sample_offset * fir_RES >> FIXP_SHIFT;
int fir_offset_rmd = sample_offset * fir_RES & FIXP_MASK;
int fir_start = (fir_offset * fir_N);
int sample_start = (sample_index - fir_N + RINGSIZE);
// Convolution with filter impulse response.
int v1 = 0;
for (int j = 0; j < fir_N; j++)
{
v1 += sample[sample_start + j] * fir[fir_start + j];
}
// Use next FIR table, wrap around to first FIR table using
// previous sample.
if (++fir_offset == fir_RES)
{
fir_offset = 0;
--sample_start;
}
fir_start = (fir_offset * fir_N);
// Convolution with filter impulse response.
int v2 = 0;
for (int j = 0; j < fir_N; j++)
{
v2 += sample[sample_start + j] * fir[fir_start + j];
}
// Linear interpolation.
// fir_offset_rmd is equal for all samples, it can thus be
// factorized out:
// sum(v1 + rmd*(v2 - v1)) = sum(v1) + rmd*(sum(v2) - sum(v1))
int v = v1 + (fir_offset_rmd * (v2 - v1) >> FIXP_SHIFT);
v >>= FIR_SHIFT;
// Saturated arithmetics to guard against 16 bit sample overflow.
int half = 1 << 15;
if (v >= half)
{
v = half - 1;
}
else if (v < -half)
{
v = -half;
}
buf[s++ *interleave] = (short)v;
}
for (int i = 0; i < delta_t.delta_t; i++)
{
clock();
sample[sample_index] = sample[sample_index + RINGSIZE] = (short)output();
++sample_index;
sample_index &= 0x3fff;
}
sample_offset -= delta_t.delta_t << FIXP_SHIFT;
delta_t.delta_t = 0;
return(s);
}