[model] cleanup iterate over output channels instead
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0bcd3b7822
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3 changed files with 27 additions and 29 deletions
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@ -6,6 +6,9 @@
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namespace Ottobit
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{
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#define BITS 16
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#define RATE 64
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void Senior::prepare(setup info)
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{
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// Initialization, this method will be called with buffer size, etc.
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@ -13,26 +16,24 @@ void Senior::prepare(setup info)
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lfo.set_sample_rate(info.rate);
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lfo.freq(5);
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skipped_frames.resize(info.input_channels);
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fms.resize(info.input_channels);
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lpfs.resize(info.input_channels);
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skipped_frames.resize(info.output_channels);
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fms.resize(info.output_channels);
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lpfs.resize(info.output_channels);
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phases.resize(info.output_channels);
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for (auto [fm, lpf] : std::views::zip(fms, lpfs))
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float init_phase{0};
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const float phase_frac{1.f / info.output_channels};
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for (auto [fm, lpf, ph] : std::views::zip(fms, lpfs, phases))
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{
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fm.set_sample_rate(info.rate);
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fm.maxDelay(0.1);
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lpf.set_sample_rate(info.rate);
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lpf.type(gam::LOW_PASS);
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}
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phases.resize(info.output_channels);
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float phase_frac{1.f / info.output_channels};
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for (int i{0}; i < info.output_channels; i++)
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{
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phases[i] = i * phase_frac;
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ph = init_phase;
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init_phase += phase_frac;
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}
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}
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@ -67,11 +68,11 @@ void Senior::operator()(tick t)
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previous_am_fm = inputs.am_fm;
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}
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// Process the input buffer
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for(int i{0}; i < inputs.audio.channels; i++)
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// Process the input buffer for each ouput channels
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for(int i{0}; i < outputs.audio.channels; i++)
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{
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auto* in = inputs.audio[i];
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auto* out = outputs.audio[i % outputs.audio.channels];
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auto* out{outputs.audio[i]};
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auto* in{inputs.audio[i % inputs.audio.channels]};
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// Init current_frame and out_frame for each channel
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double current_frame{crush(in[0])};
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@ -109,7 +110,7 @@ void Senior::operator()(tick t)
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if (inputs.filter < 1.f)
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{
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lpfs[i].freq(inputs.filter * 2920. + 80.);
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lpfs[i].freq(inputs.filter * 3000.);
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out[j] = lpfs[i](out_frame);
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}
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else
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@ -120,18 +121,18 @@ void Senior::operator()(tick t)
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void Senior::set_bit_factor()
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{
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bit_factor = pow(2, inputs.bits) - 1;
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bit_factor = pow(2, inputs.bits * BITS) - 1;
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}
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double Senior::crush(const double& f)
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{
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if (inputs.bits == BITS)
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return f;
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else
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if (inputs.bits < 1.f)
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// Rounding using static_cast
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// return static_cast<double>(static_cast<int>(f * bit_factor)) / bit_factor;
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// Close, but prefer the use of roundMagic from Lance Putnam's Gama
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return gam::scl::round<double>(f * bit_factor) / bit_factor;
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else
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return f;
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}
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} // namespace Ottobit
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@ -17,9 +17,6 @@ namespace Ottobit
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{
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class Senior
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{
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#define BITS 16
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#define RATE 64
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public:
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halp_meta(name, "Senior")
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halp_meta(category, "Audio")
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@ -32,10 +29,10 @@ public:
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{
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halp::dynamic_audio_bus<"Input", double> audio;
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halp::knob_f32<"SAMPLE RATE", halp::range{.min = .1, .max = 1., .init = 1.}> s_rate;
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halp::knob_f32<"FILTER", halp::range{.min = 0., .max = 1., .init = 1.}> filter;
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halp::knob_f32<"FILTER", halp::range{.min = .001, .max = 1., .init = 1.}> filter;
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using log_map = halp::log_mapper<std::ratio<95, 100>>;
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struct : halp::knob_f32<"BITS", halp::range{.min = 3, .max = BITS, .init = BITS}>
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struct : halp::knob_f32<"BITS", halp::range{.min = .1, .max = 1., .init = 1.}>
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{
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using mapper = log_map;
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} bits;
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@ -69,8 +66,8 @@ private:
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int bit_factor, step_size{0};
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double
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previous_bits{BITS},
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previous_rate{1},
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previous_bits{1},
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previous_depth{0},
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previous_am_fm{0},
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am_amp{0},
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@ -16,9 +16,9 @@ struct Senior::ui
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struct
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{
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halp_meta(layout, hbox)
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halp::item<&ins::bits> bits;
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halp::item<&ins::filter> filter;
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halp::item<&ins::s_rate> s_rate;
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halp::item<&ins::filter> filter;
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halp::item<&ins::bits> bits;
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} res;
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struct
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