|
// Fill out your copyright notice in the Description page of Project Settings.
|
|
|
|
#include "RenderWaveform.h"
|
|
|
|
// KISS Headers, that we need for the decompression part
|
|
#include "ThirdParty/Kiss_FFT/kiss_fft129/kiss_fft.h"
|
|
#include "ThirdParty/Kiss_FFT/kiss_fft129/tools/kiss_fftnd.h"
|
|
|
|
bool bNormalizeOutputToDb = false;
|
|
bool bShowLogDebug = false;
|
|
bool bShowWarningDebug = false;
|
|
bool bShowErrorDebug = false;
|
|
|
|
// Log Category
|
|
DECLARE_LOG_CATEGORY_EXTERN(LogRenderWave, Log, All);
|
|
|
|
// Short Defines to faster debug
|
|
#define PrintLog(TextToLog) if(bShowLogDebug) UE_LOG(LogRenderWave, Log, TextToLog)
|
|
#define PrintWarning(TextToLog) if(bShowWarningDebug) UE_LOG(LogRenderWave, Warning, TextToLog)
|
|
#define PrintError(TextToLog) if(bShowErrorDebug) UE_LOG(LogRenderWave, Error, TextToLog)
|
|
|
|
#include "Sound/SoundWave.h"
|
|
#include "AudioDevice.h"
|
|
#include "Runtime/Engine/Public/VorbisAudioInfo.h"
|
|
#include "Developer/TargetPlatform/Public/Interfaces/IAudioFormat.h"
|
|
|
|
DEFINE_LOG_CATEGORY(LogRenderWave);
|
|
|
|
float GetFFTInValue(const int16 InSampleValue, const int16 InSampleIndex, const int16 InSampleCount)
|
|
{
|
|
float FFTValue = InSampleValue;
|
|
|
|
// Apply the Hann window
|
|
FFTValue *= 0.5f * (1 - FMath::Cos(2 * PI * InSampleIndex / (InSampleCount - 1)));
|
|
|
|
return FFTValue;
|
|
}
|
|
|
|
void CalculateFrequencySpectrum(USoundWave* InSoundWaveRef, const float InStartTime, const float InDuration, TArray<float>& OutFrequencies)
|
|
{
|
|
// Clear the Array before continuing
|
|
OutFrequencies.Empty();
|
|
|
|
const int32 NumChannels = InSoundWaveRef->NumChannels;
|
|
const int32 SampleRate = InSoundWaveRef->SampleRate;
|
|
|
|
// Make sure the Number of Channels is correct
|
|
if (NumChannels > 0 && NumChannels <= 2)
|
|
{
|
|
// Check if we actually have a Buffer to work with
|
|
if (InSoundWaveRef->CachedRealtimeFirstBuffer)
|
|
{
|
|
// The first sample is just the StartTime * SampleRate
|
|
int32 FirstSample = SampleRate * InStartTime;
|
|
|
|
// The last sample is the SampleRate times (StartTime plus the Duration)
|
|
int32 LastSample = SampleRate * (InStartTime + InDuration);
|
|
|
|
// Get Maximum amount of samples in this Sound
|
|
const int32 SampleCount = InSoundWaveRef->RawPCMDataSize / (2 * NumChannels);
|
|
|
|
// An early check if we can create a Sample window
|
|
FirstSample = FMath::Min(SampleCount, FirstSample);
|
|
LastSample = FMath::Min(SampleCount, LastSample);
|
|
|
|
// Actual amount of samples we gonna read
|
|
int32 SamplesToRead = LastSample - FirstSample;
|
|
|
|
if (SamplesToRead < 0) {
|
|
|
|
PrintError(TEXT("Number of SamplesToRead is < 0!"));
|
|
return;
|
|
}
|
|
|
|
// Shift the window enough so that we get a PowerOfTwo. FFT works better with that
|
|
int32 PoT = 2;
|
|
|
|
while (SamplesToRead > PoT) {
|
|
PoT *= 2;
|
|
}
|
|
|
|
// Now we have a good PowerOfTwo to work with
|
|
SamplesToRead = PoT;
|
|
|
|
// Create two 2-dim Arrays for complex numbers | Buffer and Output
|
|
kiss_fft_cpx* Buffer[2] = {0};
|
|
kiss_fft_cpx* Output[2] = {0};
|
|
|
|
// Create 1-dim Array with one slot for SamplesToRead
|
|
int32 Dims[1] = {SamplesToRead};
|
|
|
|
// alloc once and forget, should probably move to a init/deinit func
|
|
static kiss_fftnd_cfg STF = kiss_fftnd_alloc(Dims, 1, 0, nullptr, nullptr);
|
|
|
|
int16* SamplePtr = reinterpret_cast<int16*>(InSoundWaveRef->CachedRealtimeFirstBuffer);
|
|
|
|
// Allocate space in the Buffer and Output Arrays for all the data that FFT returns
|
|
for (int32 ChannelIndex = 0; ChannelIndex < NumChannels; ChannelIndex++)
|
|
{
|
|
Buffer[ChannelIndex] = (kiss_fft_cpx*)KISS_FFT_MALLOC(sizeof(kiss_fft_cpx) * SamplesToRead);
|
|
Output[ChannelIndex] = (kiss_fft_cpx*)KISS_FFT_MALLOC(sizeof(kiss_fft_cpx) * SamplesToRead);
|
|
}
|
|
|
|
// Shift our SamplePointer to the Current "FirstSample"
|
|
SamplePtr += FirstSample * NumChannels;
|
|
|
|
float precomputeMultiplier = 2.f * PI / (SamplesToRead - 1);
|
|
|
|
for (int32 SampleIndex = 0; SampleIndex < SamplesToRead; SampleIndex++)
|
|
{
|
|
float rMult = 0.f;
|
|
if (SamplePtr != NULL && (SampleIndex + FirstSample < SampleCount))
|
|
{
|
|
rMult = 0.5f * (1.f - FMath::Cos(precomputeMultiplier * SampleIndex));
|
|
}
|
|
for (int32 ChannelIndex = 0; ChannelIndex < NumChannels; ChannelIndex++)
|
|
{
|
|
// Make sure the Point is Valid and we don't go out of bounds
|
|
if (SamplePtr != NULL && (SampleIndex + FirstSample < SampleCount))
|
|
{
|
|
// Use Window function to get a better result for the Data (Hann Window)
|
|
Buffer[ChannelIndex][SampleIndex].r = rMult * (*SamplePtr);
|
|
}
|
|
else
|
|
{
|
|
Buffer[ChannelIndex][SampleIndex].r = 0.f;
|
|
}
|
|
Buffer[ChannelIndex][SampleIndex].i = 0.f;
|
|
|
|
// Take the next Sample
|
|
SamplePtr++;
|
|
}
|
|
}
|
|
|
|
// Now that the Buffer is filled, use the FFT
|
|
for (int32 ChannelIndex = 0; ChannelIndex < NumChannels; ChannelIndex++)
|
|
{
|
|
if (Buffer[ChannelIndex])
|
|
{
|
|
kiss_fftnd(STF, Buffer[ChannelIndex], Output[ChannelIndex]);
|
|
}
|
|
}
|
|
|
|
OutFrequencies.AddZeroed(SamplesToRead);
|
|
|
|
for (int32 SampleIndex = 0; SampleIndex < SamplesToRead; ++SampleIndex)
|
|
{
|
|
float ChannelSum = 0.0f;
|
|
|
|
for (int32 ChannelIndex = 0; ChannelIndex < NumChannels; ++ChannelIndex)
|
|
{
|
|
if (Output[ChannelIndex])
|
|
{
|
|
// With this we get the actual Frequency value for the frequencies from 0hz to ~22000hz
|
|
ChannelSum += FMath::Sqrt(FMath::Square(Output[ChannelIndex][SampleIndex].r) + FMath::Square(Output[ChannelIndex][SampleIndex].i));
|
|
}
|
|
}
|
|
|
|
if (bNormalizeOutputToDb)
|
|
{
|
|
OutFrequencies[SampleIndex] = FMath::LogX(10, ChannelSum / NumChannels) * 10;
|
|
} else
|
|
{
|
|
OutFrequencies[SampleIndex] = ChannelSum / NumChannels;
|
|
}
|
|
}
|
|
|
|
// Make sure to free up the FFT stuff
|
|
// KISS_FFT_FREE(STF);
|
|
|
|
for (int32 ChannelIndex = 0; ChannelIndex < NumChannels; ++ChannelIndex)
|
|
{
|
|
KISS_FFT_FREE(Buffer[ChannelIndex]);
|
|
KISS_FFT_FREE(Output[ChannelIndex]);
|
|
}
|
|
} else {
|
|
PrintError(TEXT("InSoundVisData.PCMData is a nullptr!"));
|
|
}
|
|
} else {
|
|
PrintError(TEXT("Number of Channels is < 0!"));
|
|
}
|
|
}
|
|
|
|
void URenderWaveform::BP_RenderWaveform(USoundWave* InSoundWaveRef, UProceduralMeshComponent* Mesh, float InSongPosition, int SizeX){
|
|
if (!IsValid(InSoundWaveRef)){
|
|
return;
|
|
}
|
|
if (!IsValid(Mesh)){
|
|
return;
|
|
}
|
|
|
|
int nbVert = Mesh->GetProcMeshSection(0)->ProcVertexBuffer.Num();
|
|
bool valid;
|
|
|
|
TArray<FVector> Vertices;
|
|
TArray<FVector> Normals;
|
|
TArray<FVector2D> UV0;
|
|
TArray<FLinearColor> VertexColors;
|
|
TArray<FProcMeshTangent> Tangents;
|
|
|
|
Vertices.AddDefaulted(nbVert);
|
|
Normals.Init(FVector(0.0f, 0.0f, 1.0f), nbVert);
|
|
UV0.AddDefaulted(nbVert);
|
|
VertexColors.AddDefaulted(nbVert);
|
|
Tangents.Init(FProcMeshTangent(1.0f, 0.0f, 0.0f), nbVert);
|
|
|
|
for (size_t i = 0; i < 320; ++i){
|
|
float duration = (1 / 64.f);
|
|
float startTime = duration * i + InSongPosition;
|
|
|
|
valid = true;
|
|
if (startTime < 0.0f || startTime >= InSoundWaveRef->Duration || startTime + duration >= InSoundWaveRef->Duration) {
|
|
valid = false;
|
|
}
|
|
|
|
TArray<float> results;
|
|
|
|
if (valid) CalculateFrequencySpectrum(InSoundWaveRef, startTime, duration, results);
|
|
|
|
for (size_t j = 0; j < 64; ++j){
|
|
float height;
|
|
|
|
if (valid) height = results[j * 8.f] / 50000.f;
|
|
else height = 0;
|
|
|
|
Vertices[To1D(i, j, SizeX)] = FVector(i, j, height);
|
|
VertexColors[To1D(i, j, SizeX)] = FLinearColor(height, 0.0f, 0.0f);
|
|
}
|
|
}
|
|
|
|
Mesh->UpdateMeshSection_LinearColor(0, Vertices, Normals, UV0, VertexColors, Tangents);
|
|
|
|
return;
|
|
}
|
|
|
|
void URenderWaveform::BP_GenerateSpectrogramMesh(UProceduralMeshComponent* Mesh, int SizeX, int SizeY)
|
|
{
|
|
if (!IsValid(Mesh) || SizeX <= 0 || SizeY <= 0) {
|
|
return;
|
|
}
|
|
|
|
TArray<FVector> Vertices;
|
|
TArray<int> Faces;
|
|
TArray<FVector> Normals;
|
|
TArray<FVector2D> UV0;
|
|
TArray<FLinearColor> VertexColors;
|
|
TArray<FProcMeshTangent> Tangents;
|
|
|
|
Vertices.AddDefaulted(SizeX * SizeY);
|
|
Normals.AddDefaulted(SizeX * SizeY);
|
|
UV0.AddDefaulted(SizeX * SizeY);
|
|
VertexColors.AddDefaulted(SizeX * SizeY);
|
|
Tangents.AddDefaulted(SizeX * SizeY);
|
|
Faces.AddZeroed((SizeX - 1) * (SizeY - 1) * 6);
|
|
|
|
for (int j = 0; j < SizeY; ++j)
|
|
{
|
|
for (int i = 0; i < SizeX; ++i)
|
|
{
|
|
Vertices[To1D(i, j, SizeX)] = FVector(i,j, 0.0f);
|
|
Normals[To1D(i, j, SizeX)] = FVector(0.0f, 0.0f, 1.0f);
|
|
UV0[To1D(i, j, SizeX)] = FVector2D(0.0f, 0.0f);
|
|
VertexColors[To1D(i, j, SizeX)] = FLinearColor(0.0f, 0.0f, 0.0f);
|
|
Tangents[To1D(i, j, SizeX)] = FProcMeshTangent(1.0f, 0.0f, 0.0f);
|
|
}
|
|
}
|
|
|
|
for (int j = 0; j < SizeY - 1; ++j)
|
|
{
|
|
for (int i = 0; i < SizeX - 1; ++i)
|
|
{
|
|
Faces[To1D(i, j, SizeX - 1) * 6] = To1D(i, j, SizeX);
|
|
Faces[To1D(i, j, SizeX - 1) * 6 + 1] = To1D(i, j + 1, SizeX);
|
|
Faces[To1D(i, j, SizeX - 1) * 6 + 2] = To1D(i + 1, j, SizeX);
|
|
|
|
Faces[To1D(i, j, SizeX - 1) * 6 + 3] = To1D(i + 1, j, SizeX);
|
|
Faces[To1D(i, j, SizeX - 1) * 6 + 4] = To1D(i, j + 1, SizeX);
|
|
Faces[To1D(i, j, SizeX - 1) * 6 + 5] = To1D(i + 1, j + 1, SizeX);
|
|
}
|
|
}
|
|
|
|
Mesh->CreateMeshSection_LinearColor(0, Vertices, Faces, Normals, UV0, VertexColors, Tangents, false);
|
|
}
|
|
|
|
int URenderWaveform::To1D(int x, int y, int sizeX)
|
|
{
|
|
return (sizeX * y) + x;
|
|
}
|