Bump version -> 0.1.4

Optimize audio level loop code

.github/workflows/build.yml

@@ -24,6 +24,18 @@ jobs:
         cmake --build . --config $BUILD_TYPE -j4
         cpack
 
+    - name: Upload binary
+      uses: actions/upload-artifact@v1
+      with:
+        name: neural_amp_modeler.lv2-linux-amd64
+        path: ${{github.workspace}}/build/neural_amp_modeler.lv2
+
+#    - name: Upload deb
+#      uses: actions/upload-artifact@v1
+#      with:
+#        name: neural_amp_modeler.lv2-linux-deb-amd64
+#        path: ${{github.workspace}}/build/*.deb
+        
   build-windows:
     name: Build Windows
     runs-on: windows-latest
@@ -35,5 +47,11 @@ jobs:
     - name: Build Plugin
       working-directory: ${{github.workspace}}/build
       run: |
-        cmake.exe -G "Visual Studio 17 2022" -A x64 ..
+        cmake.exe -G "Visual Studio 17 2022" -A x64  -T ClangCL ..
         cmake --build . --config=release -j4
+        
+    - name: Upload artifact
+      uses: actions/upload-artifact@v1
+      with:
+        name: neural_amp_modeler.lv2-win
+        path: ${{github.workspace}}/build/neural_amp_modeler.lv2

.gitmodules

@@ -8,3 +8,6 @@
 	path = deps/NeuralAmpModelerCore
 	url = https://github.com/mikeoliphant/NeuralAmpModelerCore
 	branch = devel
+[submodule "deps/AudioDSPTools"]
+	path = deps/AudioDSPTools
+	url = https://github.com/sdatkinson/AudioDSPTools

CMakeLists.txt

@@ -1,10 +1,10 @@
 cmake_minimum_required(VERSION 3.10)
 
-project(NeuralAmpModelerLv2 VERSION 0.1.1)
+project(NeuralAmpModelerLv2 VERSION 0.1.4)
 
 set(CMAKE_MODULE_PATH "${CMAKE_SOURCE_DIR}/cmake")
 
-set(CMAKE_CXX_STANDARD 20)
+set(CMAKE_CXX_STANDARD 17)
 set(CMAKE_CXX_STANDARD_REQUIRED OFF)
 set(CMAKE_CXX_EXTENSIONS OFF)
 
@@ -22,11 +22,12 @@ set(NAM_LV2_ID http://github.com/mikeoliphant/neural-amp-modeler-lv2)
 
 include_directories(SYSTEM deps/eigen)
 include_directories(SYSTEM deps/lv2/include)
-include_directories(SYSTEM deps/NeuralAmpModelerCore/NAM)
+include_directories(SYSTEM deps/NeuralAmpModelerCore)
 include_directories(SYSTEM deps/json)
 include_directories(SYSTEM deps/denormal)
 
 add_definitions(-DNAM_SAMPLE_FLOAT)
+add_definitions(-DDSP_SAMPLE_FLOAT)
 
 add_subdirectory(src)
 

README.md

@@ -2,7 +2,8 @@
 
 Bare-bones implementation of [Neural Amp Modeler](https://github.com/sdatkinson/neural-amp-modeler) (NAM) models in an LV2 plugin.
 
-**There is no user interface**. Setting the model to use requires that your LV2 host supports atom:Path parameters. Reaper does not. Carla and Ardour do. If your favorite LV2 host does not support atom:Path, let them know you want it. **A Reaper feature request for this is [here](https://forum.cockos.com/showthread.php?p=2505988)**.
+**There is no user interface**. Setting the model to use requires that your LV2 host supports atom:Path parameters. Reaper does as of v6.82. Carla and Ardour do. If your favorite LV2 host does not support atom:Path, let them know you want it.
+If you are looking for a GUI version, @brummer10 [has one here](https://github.com/brummer10/neural-amp-modeler-ui) that works for Linux and Windows. You may also be interested in the the version shipped with the [MOD Desktop App](https://github.com/moddevices/mod-desktop-app), or my digital pedalboard app [Stompbox](https://github.com/mikeoliphant/StompboxUI).
 
 To get the intended behavior, **you must run your audio host at the same sample rate the model was trained at** (usually 48kHz) - no resampling is done by the plugin.
 
@@ -44,3 +45,6 @@ Note - you'll have to change the Visual Studio version if you are using a differ
 
 After building, the plugin will be in **build/neural_amp_modeler.lv2**.
 
+### Optimization
+
+If you have a relatively modern x64 processor, you can pass "**‑DUSE_NATIVE_ARCH=ON**" on your cmake command line to enable certain processor-specific optimizations.

resources/neural_amp_modeler.ttl.in

@@ -39,11 +39,11 @@
 	opts:supportedOption <http://lv2plug.in/ns/ext/buf-size#maxBlockLength>;
 
 	rdfs:comment """
-An LV2 implementation of Neural Amp Modeler.
+LV2 plugin for Neural Amp Modeler machine learning guitar amplifier simulation models
 
-NAM accepts Neural Amp Modeler .nam files.
+Based on the Neural Amp Modeler Core codebase: https://github.com/sdatkinson/NeuralAmpModelerCore
 
-A large collection of .nam files is available at https://tonehunt.org/
+A large collection of .nam models is available at https://tonehunt.org
 """;
 
 	patch:writable <@NAM_LV2_ID@#model>;

src/CMakeLists.txt

@@ -24,11 +24,18 @@ source_group(NAM ${CMAKE_CURRENT_SOURCE_DIR} FILES ${NAM_SOURCES})
 
 option(DISABLE_DENORMALS "Disable floating point denormals" ON)
 
-if(DISABLE_DENORMALS)
+if (DISABLE_DENORMALS)
 	add_definitions(-DDISABLE_DENORMALS)
-endif(DISABLE_DENORMALS)
+endif (DISABLE_DENORMALS)
 
-target_compile_features(neural_amp_modeler PUBLIC cxx_std_17)
+if (CMAKE_SYSTEM_PROCESSOR MATCHES "(amd64)|(AMD64)|(x86_64)")
+	option(USE_NATIVE_ARCH "Enable architecture-specific optimizations" OFF)
+
+	if (USE_NATIVE_ARCH)
+		set(CMAKE_CXX_FLAGS "${CMAKE_CXX_FLAGS} -march=x86-64-v3")
+		message("Enabling -march=x86-64-v3")
+	endif (USE_NATIVE_ARCH)
+endif ()
 
 set_target_properties(neural_amp_modeler
 	PROPERTIES

src/nam_lv2.cpp

@@ -27,7 +27,7 @@ static LV2_Handle instantiate(const LV2_Descriptor*, double rate, const char*, c
 
 		return nullptr;
 	}
-	catch(const std::exception)
+	catch(const std::exception&)
 	{
 		return nullptr;
 	}
@@ -96,7 +96,7 @@ LV2_SYMBOL_EXPORT const LV2_Descriptor* lv2_descriptor(uint32_t index)
 {
 	if (index == 0) {
 		// Turn on fast tanh approximation
-		activations::Activation::enable_fast_tanh();
+		nam::activations::Activation::enable_fast_tanh();
 
 		return &descriptor;
 	}

src/nam_plugin.cpp

@@ -1,10 +1,10 @@
 #include <algorithm>
 #include <cmath>
 #include <utility>
+#include <cassert>
 
 #include "nam_plugin.h"
-#include "activations.h"
-#include <cassert>
+#include <NAM/activations.h>
 
 #define SMOOTH_EPSILON .0001f
 
@@ -20,8 +20,10 @@ namespace NAM {
 		delete currentModel;
 	}
 
-	bool Plugin::initialize(double rate, const LV2_Feature* const* features) noexcept
+	bool Plugin::initialize(double sampleRate, const LV2_Feature* const* features) noexcept
 	{
+		this->sampleRate = sampleRate;
+
 		// for fetching initial options, can be null
 		LV2_Options_Option* options = nullptr;
 
@@ -86,7 +88,7 @@ namespace NAM {
 				auto msg = static_cast<const LV2LoadModelMsg*>(data);
 				auto nam = static_cast<NAM::Plugin*>(instance);
 
-				::DSP* model = nullptr;
+				nam::DSP* model = nullptr;
 				LV2SwitchModelMsg response = { kWorkTypeSwitch, {}, {} };
 				LV2_Worker_Status result = LV2_WORKER_SUCCESS;
 
@@ -105,18 +107,15 @@ namespace NAM {
 					{
 						lv2_log_trace(&nam->logger, "Staging model change: `%s`\n", msg->path);
 
-						model = get_dsp(msg->path).release();
-
-						// Enable model loudness normalization
-						model->SetNormalize(true);
+						model = nam::get_dsp(msg->path).release();
 
 						// Pre-run model to ensure all needed buffers are allocated in advance
 						if (const int32_t numSamples = nam->maxBufferSize)
 						{
 							float* buffer = new float[numSamples];
+							memset(buffer, 0, numSamples * sizeof(float));
 
-							std::unordered_map<std::string, double> params = {};
-							model->process(&buffer, &buffer, 1, numSamples, 1.0, 1.0, params);
+							model->process(buffer, buffer, numSamples);
 							model->finalize_(numSamples);
 
 							delete[] buffer;
@@ -127,7 +126,7 @@ namespace NAM {
 
 					memcpy(response.path, msg->path, pathlen);
 				}
-				catch (std::exception)
+				catch (const std::exception&)
 				{
 					response.path[0] = '\0';
 
@@ -220,57 +219,87 @@ namespace NAM {
 			}
 		}
 
+		float level;
+
 		// convert input level from db
 		float desiredInputLevel = powf(10, *(ports.input_level) * 0.05f);
 
 		if (fabs(desiredInputLevel - inputLevel) > SMOOTH_EPSILON)
 		{
+			level = inputLevel;
 			for (unsigned int i = 0; i < n_samples; i++)
 			{
 				// do very basic smoothing
-				inputLevel = (.99f * inputLevel) + (.01f * desiredInputLevel);
+				level = (.99f * level) + (.01f * desiredInputLevel);
 
-				ports.audio_out[i] = ports.audio_in[i] * inputLevel;
+				ports.audio_out[i] = ports.audio_in[i] * level;
 			}
+
+			inputLevel = level;
 		}
 		else
 		{
-			inputLevel = desiredInputLevel;
+			level = inputLevel = desiredInputLevel;
 
 			for (unsigned int i = 0; i < n_samples; i++)
 			{
-				ports.audio_out[i] = ports.audio_in[i] * inputLevel;
+				ports.audio_out[i] = ports.audio_in[i] * level;
 			}
 		}
 
+		float modelLoudnessAdjustmentDB = 0;
+
 		if (currentModel != nullptr)
 		{
-			currentModel->process(&ports.audio_out, &ports.audio_out, 1, n_samples, 1.0, 1.0, mNAMParams);
+			currentModel->process(ports.audio_out, ports.audio_out, n_samples);
 			currentModel->finalize_(n_samples);
+
+			if (currentModel->HasLoudness())
+			{
+				// Normalize model to -18dB
+				modelLoudnessAdjustmentDB = -18 - currentModel->GetLoudness();
+			}
 		}
 
-		// convert output level from db
-		float desiredOutputLevel = powf(10, *(ports.output_level) * 0.05f);
+		// Convert output level from db
+		float desiredOutputLevel = powf(10, (*(ports.output_level) + modelLoudnessAdjustmentDB) * 0.05f);
 
 		if (fabs(desiredOutputLevel - outputLevel) > SMOOTH_EPSILON)
 		{
+			level = outputLevel;
+
 			for (unsigned int i = 0; i < n_samples; i++)
 			{
 				// do very basic smoothing
-				outputLevel = (.99f * outputLevel) + (.01f * desiredOutputLevel);
+				level = (.99f * level) + (.01f * desiredOutputLevel);
 
-				ports.audio_out[i] *= outputLevel;
+				ports.audio_out[i] = ports.audio_out[i] * outputLevel;
 			}
+
+			outputLevel = level;
 		}
 		else
 		{
-			outputLevel = desiredOutputLevel;
+			level = outputLevel = desiredOutputLevel;
 
 			for (unsigned int i = 0; i < n_samples; i++)
 			{
-				ports.audio_out[i] *= outputLevel;
+				ports.audio_out[i] = ports.audio_out[i] * level;
 			}
 		}
+
+		float dcBlockCoefficient = 1 - (220.0 / sampleRate);
+
+		for (unsigned int i = 0; i < n_samples; i++)
+		{
+			float dcInput = ports.audio_out[i];
+
+			// dc blocker
+			ports.audio_out[i] = ports.audio_out[i] - prevDCInput + dcBlockCoefficient * prevDCOutput;
+
+			prevDCInput = dcInput;
+			prevDCOutput = ports.audio_out[i];
+		}
 	}
 
 	uint32_t Plugin::options_get(LV2_Handle, LV2_Options_Option*)

src/nam_plugin.h

@@ -21,7 +21,7 @@
 #include <lv2/state/state.h>
 #include <lv2/units/units.h>
 
-#include "dsp.h"
+#include <NAM/dsp.h>
 
 #define PlUGIN_URI "http://github.com/mikeoliphant/neural-amp-modeler-lv2"
 #define MODEL_URI PlUGIN_URI "#model"
@@ -43,12 +43,12 @@ namespace NAM {
 	struct LV2SwitchModelMsg {
 		LV2WorkType type;
 		char path[MAX_FILE_NAME];
-		::DSP* model;
+		nam::DSP* model;
 	};
 
 	struct LV2FreeModelMsg {
 		LV2WorkType type;
-		::DSP* model;
+		nam::DSP* model;
 	};
 
 	class Plugin {
@@ -64,14 +64,16 @@ namespace NAM {
 
 		Ports ports = {};
 
+		double sampleRate;
+
 		LV2_URID_Map* map = nullptr;
 		LV2_Log_Logger logger = {};
 		LV2_Worker_Schedule* schedule = nullptr;
 
-		::DSP* currentModel = nullptr;
+		nam::DSP* currentModel = nullptr;
 		std::string currentModelPath;
-
-		std::unordered_map<std::string, double> mNAMParams = {};
+		float prevDCInput = 0;
+		float prevDCOutput = 0;
 
 		Plugin();
 		~Plugin();