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Mike Oliphant
2023-03-09 09:10:13 -08:00
parent c3cce3fa42
commit 2eff99c3ff
5 changed files with 16 additions and 68 deletions
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resources/neural_amp_modeler.ttl.in
+4 -39
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@@ -10,22 +10,12 @@
@prefix units: <http://lv2plug.in/ns/extensions/units#>.
@prefix urid: <http://lv2plug.in/ns/ext/urid#>.
@prefix param: <http://lv2plug.in/ns/ext/parameters#>.
@prefix pg: <http://lv2plug.in/ns/ext/port-groups#>.
<@NAM_LV2_ID@>
a doap:Project;
doap:maintainer <http://github.com/mikeoliphant>;
doap:name "Neural Amp Modeler".
<@NAM_LV2_ID@#input>
a pg:MonoGroup, pg:InputGroup;
lv2:symbol "input".
<h@NAM_LV2_ID@#output>
a pg:MonoGroup, pg:OutputGroup;
lv2:symbol "output";
pg:source <@NAM_LV2_ID@#input>.
<http://github.com/mikeoliphant/neural-amp-modeler-lv2>
a lv2:Plugin, lv2:AmplifierPlugin;
doap:name "Neural Amp Modeler";
@@ -39,49 +29,24 @@
rdfs:comment "An LV2 implementation of Neural Amp Modeler";
pg:mainInput <@NAM_LV2_ID@#input>;
pg:mainOutput <@NAM_LV2_ID@#output>;
# Control Ports
# Audio Ports
lv2:port [
a lv2:InputPort, atom:AtomPort;
atom:bufferType atom:Sequence;
lv2:designation lv2:control ;
lv2:index 0;
lv2:symbol "control";
lv2:name "control";
rdfs:comment "UI -> DSP communication"
], [
a lv2:OutputPort, atom:AtomPort;
atom:bufferType atom:Sequence;
lv2:designation lv2:control ;
lv2:index 1;
lv2:symbol "notify";
lv2:name "Notify";
# amount of data sent in a single 8192 sample process block
rsz:minimumSize 131428;
rdfs:comment "DSP -> UI communication"
], [
a lv2:InputPort, lv2:AudioPort;
lv2:index 2;
lv2:index 0;
lv2:symbol "input";
lv2:name "Input";
pg:group <http://github.com/mikeoliphant/neural-amp-modeler-lv2#input>;
lv2:designation pg:left
], [
a lv2:OutputPort, lv2:AudioPort;
lv2:index 3;
lv2:index 1;
lv2:symbol "output";
lv2:name "Output";
pg:group <http://github.com/mikeoliphant/neural-amp-modeler-lv2#output>;
lv2:designation pg:left
];
# Mixer
lv2:port [
a lv2:InputPort, lv2:ControlPort;
lv2:designation param:wetDryRatio;
lv2:index 4;
lv2:index 2;
lv2:symbol "mix";
lv2:name "Mix";
rdfs:comment "dry/wet ratio";
src/dsp.cpp
+8 -12
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@@ -20,14 +20,14 @@ constexpr auto _INPUT_BUFFER_SAFETY_FACTOR = 32;
DSP::DSP() { this->_stale_params = true; }
void DSP::process(const NAMSample *input, NAMSample *output,
const int num_channels, const int num_frames,
const int num_frames,
const double input_gain, const double output_gain,
const std::unordered_map<std::string, double> &params) {
this->_get_params_(params);
this->_apply_input_level_(input, num_channels, num_frames, input_gain);
this->_apply_input_level_(input, num_frames, input_gain);
this->_ensure_core_dsp_output_ready_();
this->_process_core_();
this->_apply_output_level_(output, num_channels, num_frames, output_gain);
this->_apply_output_level_(output, num_frames, output_gain);
}
void DSP::finalize_(const int num_frames) {}
@@ -46,14 +46,12 @@ void DSP::_get_params_(
}
}
void DSP::_apply_input_level_(const NAMSample *input, const int num_channels,
const int num_frames, const double gain) {
void DSP::_apply_input_level_(const NAMSample *input, const int num_frames, const double gain) {
// Must match exactly; we're going to use the size of _input_post_gain later
// for num_frames.
if (this->_input_post_gain.size() != num_frames)
this->_input_post_gain.resize(num_frames);
// MONO ONLY
const int channel = 0;
for (int i = 0; i < num_frames; i++)
this->_input_post_gain[i] = float(gain * input[i]);
}
@@ -69,11 +67,9 @@ void DSP::_process_core_() {
this->_core_dsp_output[i] = this->_input_post_gain[i];
}
void DSP::_apply_output_level_(NAMSample *output, const int num_channels,
const int num_frames, const double gain) {
for (int c = 0; c < num_channels; c++)
for (int s = 0; s < num_frames; s++)
output[s] = NAMSample(gain * this->_core_dsp_output[s]);
void DSP::_apply_output_level_(NAMSample *output, const int num_frames, const double gain) {
for (int s = 0; s < num_frames; s++)
output[s] = NAMSample(gain * this->_core_dsp_output[s]);
}
// Buffer =====================================================================
src/dsp.h
+3 -6
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@@ -42,8 +42,7 @@ public:
// 3. The core DSP algorithm is run (This is what should probably be
// overridden in subclasses).
// 4. The output level is applied and the result stored to `output`.
virtual void process(const NAMSample *input, NAMSample *output,
const int num_channels, const int num_frames,
virtual void process(const NAMSample *input, NAMSample *output, const int num_frames,
const double input_gain, const double output_gain,
const std::unordered_map<std::string, double> &params);
// Anything to take care of before next buffer comes in.
@@ -74,8 +73,7 @@ protected:
// Apply the input gain
// Result populates this->_input_post_gain
void _apply_input_level_(const NAMSample *input, const int num_channels,
const int num_frames, const double gain);
void _apply_input_level_(const NAMSample *input, const int num_frames, const double gain);
// i.e. ensure the size is correct.
void _ensure_core_dsp_output_ready_();
@@ -86,8 +84,7 @@ protected:
virtual void _process_core_();
// Copy this->_core_dsp_output to output and apply the output volume
void _apply_output_level_(NAMSample *output, const int num_channels,
const int num_frames, const double gain);
void _apply_output_level_(NAMSample *output, const int num_frames, const double gain);
};
// Class where an input buffer is kept so that long-time effects can be
src/nam_plugin.cpp
+1 -9
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@@ -20,15 +20,7 @@ namespace NAM {
}
void Plugin::process(uint32_t n_samples) noexcept {
if (ports.control) {
LV2_ATOM_SEQUENCE_FOREACH(ports.control, event) {
if (event->body.type == uris.atom_Object) {
const auto obj = reinterpret_cast<LV2_Atom_Object*>(&event->body);
}
}
}
namModel->process(ports.audio_in, ports.audio_out, 1, n_samples, 1.0, 1.0, mNAMParams);
namModel->process(ports.audio_in, ports.audio_out, n_samples, 1.0, 1.0, mNAMParams);
namModel->finalize_(n_samples);
}
}
src/nam_plugin.hpp
-2
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@@ -20,8 +20,6 @@ namespace NAM {
static constexpr std::string_view URI = "http://github.com/mikeoliphant/neural-amp-modeler-lv2";
struct Ports {
const LV2_Atom_Sequence* control;
LV2_Atom_Sequence* notify;
const float* audio_in;
float* audio_out;
};