Removed the Player module, implemented FFT based lowpass filter and implemented FileWriter to write out filtered audio file.

Amsakan Bavan · Amsakan Bavan · commit 181673e833de · 2025-08-01T13:51:42.000+02:00
diff --git a/.idea/vcs.xml b/.idea/vcs.xml
diff --git a/CMakeLists.txt b/CMakeLists.txt
@@ -4,21 +4,12 @@ project(simpleFilter)
 set(CMAKE_CXX_STANDARD 23)
 set(CMAKE_CXX_STANDARD_REQUIRED ON)
 
-
 if(MSVC)
-    add_compile_options(/std:c++17)
+    add_compile_options(/Zc:__cplusplus)
 endif()
 
-
 include(FetchContent)
 
-FetchContent_Declare(
-        portaudio
-        GIT_REPOSITORY https://github.com/PortAudio/portaudio
-        GIT_TAG master
-)
-FetchContent_MakeAvailable(portaudio)
-
 FetchContent_Declare(
         sndfile
         GIT_REPOSITORY https://github.com/libsndfile/libsndfile
@@ -33,7 +24,16 @@ FetchContent_Declare(
 )
 FetchContent_MakeAvailable(abseil-cpp)
 
+FetchContent_Declare(
+        fftw3
+        URL https://www.fftw.org/fftw-3.3.10.tar.gz
+        URL_HASH MD5=8ccbf6a5ea78a16dbc3e1306e234cc5c
+)
+set(ENABLE_FLOAT ON CACHE BOOL "Enable single precision" FORCE)
+FetchContent_MakeAvailable(fftw3)
+
 
+include_directories("${fftw3_SOURCE_DIR}/api")
 include_directories("${CMAKE_SOURCE_DIR}/include")
 
 add_subdirectory(src)
diff --git a/include/FileWriter.hh b/include/FileWriter.hh
@@ -0,0 +1,13 @@
+//
+// Created by abav on 01.08.2025.
+//
+
+#ifndef FILEWRITER_HH
+#define FILEWRITER_HH
+#include <string>
+#include <vector>
+
+namespace FileWriter {
+    void writeFile(std::string fileName, std::vector<float> audioData);
+};
+#endif //FILEWRITER_HH
diff --git a/include/Filter.hh b/include/Filter.hh
@@ -1,12 +1,23 @@
 //
-// Created by abav on 17.07.2025.
+// Created by abav on 01.08.2025.
 //
 
 #ifndef FILTER_HH
 #define FILTER_HH
+#include <fftw3.h>
 #include <vector>
 
-namespace Filter {
-    void filter(std::vector<float>& audio, float filterFrequency);
-}
+class Filter {
+    int N;                    // Signal length
+    float* input;           // Input buffer
+    fftwf_complex* fft_data;  // FFT output buffer
+    float* output;          // Output buffer (filtered signal)
+    fftwf_plan forward_plan;  // Forward FFT plan
+    fftwf_plan inverse_plan;  // Inverse FFT plan
+
+public:
+    Filter(int signal_length);
+    ~Filter();
+    void filter(std::vector<float>& signal, double cutoff_freq, double sample_rate = 44100);
+};
 #endif //FILTER_HH
diff --git a/src/CLI/CLI.cc b/src/CLI/CLI.cc
@@ -1,24 +1,18 @@
-//
-// Created by abav on 17.07.2025.
-//
 #include "CLI/CLI.hh"
 #include "CLI/Arguments.hh"
 
-#include <iostream>
 #include <vector>
 #include <absl/strings/str_split.h>
 
-#include "FileReader.hh"
 #include "Filter.hh"
-#include "Player.hh"
+#include "FileReader.hh"
+#include "FileWriter.hh"
+
 
 void CLI::run(int argc, char** argv) {
     Arguments::CLIArguments arguments = Arguments::parseArguments(argc, argv);
     std::vector<float> audio = FileReader::readFile(arguments.fileName);
-/*
-        std::cout << "Doing filtering..\n";
-    Filter::filter(audio, arguments.filterFrequency);
-    std::cout << "Done.\n";
-*/
-    Player::play(audio);
-};
+    Filter filter(audio.size());
+    filter.filter(audio, arguments.filterFrequency);
+    FileWriter::writeFile("filtered.wav", audio);
+}
diff --git a/src/CLI/CMakeLists.txt b/src/CLI/CMakeLists.txt
@@ -5,7 +5,7 @@ add_library(CLI
 
 target_link_libraries(CLI
     FileReader
+    FileWriter
     Filter
-    Player
     absl::strings
 )
diff --git a/src/CMakeLists.txt b/src/CMakeLists.txt
@@ -1,5 +1,5 @@
 add_subdirectory(CLI)
-add_subdirectory(FileReader)
 add_subdirectory(Filter)
-add_subdirectory(Main)
-add_subdirectory(Player)
+add_subdirectory(FileReader)
+add_subdirectory(FileWriter)
+add_subdirectory(Main)
diff --git a/src/FileReader/FileReader.cc b/src/FileReader/FileReader.cc
@@ -3,11 +3,13 @@
 //
 #include "FileReader.hh"
 
+#include <iostream>
 #include <sndfile.h>
 
 std::vector<float> FileReader::readFile(std::string filePath) {
     SF_INFO fileInfo;
     SNDFILE* audioFile = sf_open(filePath.c_str(), SFM_READ, &fileInfo);
+    
     const auto fileAudio = new float[fileInfo.frames];
     sf_read_float(audioFile, fileAudio, fileInfo.frames);
     sf_close(audioFile);
diff --git a/src/FileWriter/CMakeLists.txt b/src/FileWriter/CMakeLists.txt
@@ -0,0 +1,7 @@
+add_library(FileWriter
+    FileWriter.cc
+)
+
+target_link_libraries(FileWriter
+    sndfile
+)
diff --git a/src/FileWriter/FileWriter.cc b/src/FileWriter/FileWriter.cc
@@ -0,0 +1,18 @@
+//
+// Created by abav on 01.08.2025.
+//
+#include "FileWriter.hh"
+#include <iostream>
+#include <sndfile.h>
+
+void FileWriter::writeFile(std::string fileName, std::vector<float> audioData) {
+    SF_INFO sfInfo;
+    sfInfo.channels = 1;
+    sfInfo.samplerate = 44100;
+    sfInfo.format = SF_FORMAT_WAV | SF_FORMAT_FLOAT;
+
+    SNDFILE* audioFile = sf_open(fileName.c_str(), SFM_WRITE, &sfInfo);
+    std::cout << "Writing audio files..." << std::endl;
+    sf_write_float(audioFile, audioData.data(), audioData.size());
+    sf_close(audioFile);
+}
diff --git a/src/Filter/CMakeLists.txt b/src/Filter/CMakeLists.txt
@@ -1,3 +1,7 @@
 add_library(Filter
     Filter.cc
+)
+
+target_link_libraries(Filter
+    fftw3f
 )
diff --git a/src/Filter/Filter.cc b/src/Filter/Filter.cc
@@ -1,51 +1,54 @@
-//
-// Created by abav on 17.07.2025.
-//
 #include "Filter.hh"
+#include <iostream>
+
+Filter::Filter(int signal_length) : N(signal_length) {
+    std::cout << "Setting up filter..." << std::endl;
+    // Allocate memory for FFTW arrays
+    input = fftwf_alloc_real(N);
+    fft_data = fftwf_alloc_complex(N/2 + 1);  // Real FFT only needs N/2+1 complex values
+    output = fftwf_alloc_real(N);
+
+    // Create FFTW plans
+    forward_plan = fftwf_plan_dft_r2c_1d(N, input, fft_data, FFTW_MEASURE);
+    inverse_plan = fftwf_plan_dft_c2r_1d(N, fft_data, output, FFTW_MEASURE);
+}
 
-#include <complex>
+Filter::~Filter() {
+    // Clean up FFTW resources
+    fftwf_destroy_plan(forward_plan);
+    fftwf_destroy_plan(inverse_plan);
+    fftwf_free(input);
+    fftwf_free(fft_data);
+    fftwf_free(output);
+}
 
-#ifdef _WIN32
-#include <corecrt_math_defines.h>
-#else
-#include <math.h>
-#endif
+void Filter::filter(std::vector<float>& signal, double cutoff_freq, double sample_rate) {
+    std::cout << "Filtering..." << std::endl;
+    // Copy input signal to FFTW input buffer
+    for (int i = 0; i < N; i++) {
+        input[i] = (i < signal.size()) ? signal[i] : 0.0f;
+    }
 
-void Filter::filter(std::vector<float>& audio, float filterFrequency) {
-    std::vector<std::complex<float>> frequencyBins;
-    std::complex<float> j(0, 1);
-    std::complex<float> negativeJ(0, -1);
-    int N = audio.size();
+    // Execute forward FFT
+    fftwf_execute(forward_plan);
 
-    // Forward DFT
-    for (int omega = 0; omega < N; omega++) {
-        std::complex<float> sum(0, 0);
+    // Apply lowpass filter in frequency domain
+    int cutoff_bin = static_cast<int>((cutoff_freq / sample_rate) * N);
 
-        for (int k = 0; k < N; k++) {
-            float angle = 2.0f * M_PI * omega * k / static_cast<float>(N);
-            sum += audio[k] * std::exp(negativeJ * angle);
+    for (int i = 0; i <= N/2; i++) {
+        if (i > cutoff_bin) {
+            // Zero out frequencies above cutoff
+            fft_data[i][0] = 0.0;  // Real part
+            fft_data[i][1] = 0.0;  // Imaginary part
         }
-
-        frequencyBins.push_back(sum);
     }
 
-    // Filtering - convert filterFrequency to bin index
-    float sampleRate = 44100.0f; // Assuming 44.1kHz
-    int cutoffBin = static_cast<int>(filterFrequency * N / sampleRate);
-
-    for (int i = cutoffBin; i < N; i++) {
-        frequencyBins[i] = 0;
-    }
-
-    // Inverse DFT
-    for (int n = 0; n < N; n++) {
-        std::complex<float> sum(0, 0);
-
-        for (int k = 0; k < N; k++) {
-            float angle = 2.0f * M_PI * k * n / static_cast<float>(N);
-            sum += frequencyBins[k] * std::exp(j * angle);
-        }
+    // Execute inverse FFT
+    fftwf_execute(inverse_plan);
 
-        audio[n] = sum.real() / static_cast<float>(N);
+    // Copy and normalize output (FFTW inverse doesn't normalize)
+    signal.resize(N);
+    for (int i = 0; i < N; i++) {
+        signal[i] = output[i] / N;
     }
 }
diff --git a/src/Main/Main.cc b/src/Main/Main.cc
@@ -1,7 +1,9 @@
+//
+// Created by abav on 29.07.2025.
+//
 #include "CLI/CLI.hh"
 
 int main(int argc, char** argv) {
     CLI::run(argc, argv);
-
     return 0;
-}
+}
diff --git a/src/Player/CMakeLists.txt b/src/Player/CMakeLists.txt
diff --git a/src/Player/Player.cc b/src/Player/Player.cc

Original file line number	Diff line number	Diff line change
`@@ -5,7 +5,7 @@ add_library(CLI`
`5`	`5`
`6`	`6`	`target_link_libraries(CLI`
`7`	`7`	`FileReader`
	`8`	`+ FileWriter`
`8`	`9`	`Filter`
`9`		`- Player`
`10`	`10`	`absl::strings`
`11`	`11`	`)`