Added backend support for audio buffers (PipeWire)

plugin/AudioImageProvider.cpp

@@ -0,0 +1,16 @@
+#include "AudioImageProvider.h"
+
+AudioImageProvider::AudioImageProvider() 
+    : QQuickImageProvider(QQuickImageProvider::Pixmap) {}
+
+QPixmap AudioImageProvider::requestPixmap(const QString &id, QSize *size, const QSize &requestedSize)
+{
+    Q_UNUSED(id) // id is useless here. we always want to return the latest frame from AudioModel
+    Q_UNUSED(requestedSize) // requested size is useless too. texture must always be 512x2
+
+    if(size)
+        *size = AudioModel::frame().size();
+
+    //return the latest frame
+    return AudioModel::frame();
+}

plugin/AudioImageProvider.h

@@ -0,0 +1,17 @@
+#ifndef AUDIOIMAGEPROVIDER_H
+#define AUDIOIMAGEPROVIDER_H
+#include <QObject>
+#include <QPixmap>
+#include <QQuickImageProvider>
+
+#include "AudioModel.h"
+
+class AudioImageProvider : public QQuickImageProvider
+{
+    public:
+        explicit AudioImageProvider();
+
+        QPixmap requestPixmap(const QString &id, QSize *size, const QSize &requestedSize) override;
+};
+
+#endif

plugin/AudioModel.cpp

@@ -1,72 +1,363 @@
+/* PipeWire */
+/* SPDX-FileCopyrightText: Copyright © 2022 Wim Taymans */
+/* SPDX-License-Identifier: MIT */
+
+/*
+ [title]
+ Audio capture using \ref pw_stream "pw_stream".
+ [title]
+ */
+
+#include <stdio.h>
+#include <math.h>
+#include <fftw3.h>
+
 #include "AudioModel.h"
 
-#ifdef AUDIOMODEL_H
+AudioModel::AudioModel(QObject *parent) : QObject(parent)
-
-AudioModel::AudioModel(QObject *parent)
-    : QObject(parent), m_deviceString(QString())
 {
+    m_impl_data = { nullptr, nullptr, 0, 1, {}, {}};
+    m_impl_data.samples.reserve(4096);
+    m_impl_data.smoothed.reserve(2048);
+
+    //fill the smoothed data buffer with 0s
+    for(int i = 0; i < 2048; ++i)
+        m_impl_data.smoothed.insert(i, 0);
+
+    m_thread = new QThread(parent);
+
+    moveToThread(m_thread);
+    connect(m_thread, &QThread::started, this, &AudioModel::startCaptureAsync);
+
+    const struct spa_pod *params[1];
+    uint8_t buffer[1024];
+    struct pw_properties *props;
+    struct spa_pod_builder b = SPA_POD_BUILDER_INIT(buffer, sizeof(buffer));
+
+    pw_init(nullptr, nullptr);
+
+    /* make a main loop. If you already have another main loop, you can add
+         * the fd of this pipewire mainloop to it. */
+    m_impl_data.loop = pw_main_loop_new(NULL);
+
+    pw_loop_add_signal(pw_main_loop_get_loop(m_impl_data.loop), SIGINT, do_quit, &m_impl_data);
+    pw_loop_add_signal(pw_main_loop_get_loop(m_impl_data.loop), SIGTERM, do_quit, &m_impl_data);
+
+    /* Create a simple stream, the simple stream manages the core and remote
+         * objects for you if you don't need to deal with them.
+         *
+         * If you plan to autoconnect your stream, you need to provide at least
+         * media, category and role properties.
+         *
+         * Pass your events and a user_data pointer as the last arguments. This
+         * will inform you about the stream state. The most important event
+         * you need to listen to is the process event where you need to produce
+         * the data.
+         */
+    props = pw_properties_new(PW_KEY_MEDIA_TYPE, "Audio",
+                              PW_KEY_MEDIA_CATEGORY, "Capture",
+                              PW_KEY_MEDIA_ROLE, "Music",
+                              NULL);
+
+    /* uncomment if you want to capture from the sink monitor ports */
+    pw_properties_set(props, PW_KEY_STREAM_CAPTURE_SINK, "true");
+
+    m_impl_data.stream = pw_stream_new_simple(
+        pw_main_loop_get_loop(m_impl_data.loop),
+        "audio-capture",
+        props,
+        &stream_events,
+        &m_impl_data);
+
+    struct spa_audio_info_raw info = SPA_AUDIO_INFO_RAW_INIT(
+                                        .format = SPA_AUDIO_FORMAT_F32,
+                                        .rate = 44100,
+                                        .channels = 2
+                                    );
+
+    /* Make one parameter with the supported formats. The SPA_PARAM_EnumFormat
+         * id means that this is a format enumeration (of 1 value).
+         * We leave the channels and rate empty to accept the native graph
+         * rate and channels. */
+    params[0] = spa_format_audio_raw_build(&b, SPA_PARAM_EnumFormat, &info);
+
+    /* Now connect this stream. We ask that our process function is
+         * called in a realtime thread. */
+    pw_stream_connect(m_impl_data.stream,
+                      PW_DIRECTION_INPUT,
+                      PW_ID_ANY,
+                      static_cast<pw_stream_flags>(PW_STREAM_FLAG_AUTOCONNECT |
+                          PW_STREAM_FLAG_MAP_BUFFERS |
+                          PW_STREAM_FLAG_RT_PROCESS),
+                      params, 1);
+
+    if(!m_instance)
+        m_instance = this;
 }
 
 AudioModel::~AudioModel()
 {
-    if (m_recorder) 
+    stopCapture();
+
+    if (m_impl_data.stream)
     {
-        m_recorder->stop();
+        pw_stream_disconnect(m_impl_data.stream);
-        delete m_recorder;
+        pw_stream_destroy(m_impl_data.stream);
+    }
+    if (m_impl_data.loop)
+        pw_main_loop_destroy(m_impl_data.loop);
+
+    if(m_thread)
+    {
+        if(m_thread->isRunning())
+            m_thread->quit();
+
+        m_thread->deleteLater();
     }
 
-    if (m_audioInput) 
+    pw_deinit();
-        delete m_audioInput;
-
-    if (m_captureSession)
-        delete m_captureSession;
 }
 
-QByteArray AudioModel::frame() const
+void AudioModel::startCapture()
 {
-    // This function should return the current audio frame.
+    if(m_thread->isRunning())
-    // For now, we return an empty QByteArray.
-    return QByteArray();
-}
-
-QString AudioModel::device() const
-{
-    return m_deviceString;
-}
-
-QStringList AudioModel::availableDevices() const
-{
-    QStringList devices;
-    
-    // Assuming QAudioDeviceInfo is used to get available audio devices
-    for (const auto &device : QMediaDevices::audioInputs()) 
-    {
-        devices.append(QString::fromLatin1(device.id()));
-    }
-
-    return devices;
-}
-
-void AudioModel::setDeviceName(const QString &device)
-{
-    if (m_deviceString == device)
         return;
 
-    m_deviceString = device;
+    m_thread->start(QThread::NormalPriority);
-
-    // if (m_audioInput) 
-    // {
-    //     m_audioInput->setDevice(QAudioInput(device));
-    //     getAudioFrame();
-    // }
 }
 
-void AudioModel::getAudioFrame()
+void AudioModel::stopCapture()
 {
-    // This function should be implemented to retrieve the audio frame
+    m_running = false;
-    // from the audio input device and emit the frameChanged signal.
+    pw_main_loop_quit(m_impl_data.loop);
-    // For now, we will just emit the signal to indicate that the frame is ready.
-    Q_EMIT frameChanged();
 }
 
-#endif
+void AudioModel::startCaptureAsync()
+{
+    pw_main_loop_run(m_impl_data.loop);
+}
+
+QPixmap AudioModel::frame()
+{
+    return m_instance->m_frame;
+}
+
+/* Be notified when the stream param changes. We're only looking at the
+ * format changes.
+ */
+void AudioModel::on_stream_param_changed(void *_data, uint32_t id, const struct spa_pod *param)
+{
+    struct impl *data = reinterpret_cast<impl*>(_data);
+
+    /* NULL means to clear the format */
+    if (param == NULL || id != SPA_PARAM_Format)
+        return;
+
+    if (spa_format_parse(param, &data->format.media_type, &data->format.media_subtype) < 0)
+        return;
+
+    /* only accept raw audio */
+    if (data->format.media_type != SPA_MEDIA_TYPE_audio ||
+        data->format.media_subtype != SPA_MEDIA_SUBTYPE_raw)
+        return;
+
+    /* call a helper function to parse the format for us. */
+    spa_format_audio_raw_parse(param, &data->format.info.raw);
+
+    fprintf(stdout, "capturing rate:%d channels:%d\n", data->format.info.raw.rate, data->format.info.raw.channels);
+}
+
+/* our data processing function is in general:
+ *
+ *  struct pw_buffer *b;
+ *  b = pw_stream_dequeue_buffer(stream);
+ *
+ *  .. consume stuff in the buffer ...
+ *
+ *  pw_stream_queue_buffer(stream, b);
+ */
+void AudioModel::on_process(void *userdata)
+{
+    struct impl *data = reinterpret_cast<impl*>(userdata);
+    struct pw_buffer *b;
+    struct spa_buffer *buf;
+
+    float *samples, max;
+    uint32_t c, n, n_channels, n_samples, peak;
+
+    if ((b = pw_stream_dequeue_buffer(data->stream)) == NULL) {
+        pw_log_warn("out of buffers: %m");
+        return;
+    }
+
+    buf = b->buffer;
+    if ((samples = reinterpret_cast<float*>(buf->datas[0].data)) == NULL)
+        return;
+
+    n_channels = data->format.info.raw.channels;
+    n_samples = buf->datas[0].chunk->size / sizeof(float);
+
+    // convert channels to mono
+    for(int index = 0; index < n_samples; index += n_channels)
+    {
+        float average = 0;
+
+        for(int channel = 0; channel < n_channels; channel++)
+            average += samples[index + channel];
+
+        average /= n_channels;
+
+        if(index > 0)
+            data->samples.push_back(average);
+    }
+
+    /**
+     * To convert the captured samples to an audio texture we need to:
+     *
+     * Take 2048 samples of audio data as an array of floating point data
+     * 1. Calculate wave data
+     * 2. Multiply it with Blackman window
+     * 3. Convert samples into complex numbers (imaginary parts are all zeros)
+     * 4. Apply the Fourier transform with fftSize = 2048, as a result we get 1024 FFT bins
+     * 5. Convert complex result into real values using cabs() function
+     * 6. Divide each value by fftSize
+     * 7. Apply smoothing by using previously calculated spectrum values
+     * 8. Convert resulting values to dB: dB = 20 * log10(v)
+     * 9. Convert floating point dB spectrum into 8-bit values:
+     * 10. Write 8-bit values into texture
+     */
+
+    // 1
+    if(data->samples.length() >= 2048)
+    {
+        QVector<qreal> rawSamples = data->samples.mid(0, 2048);
+        data->samples.remove(0, 2048);
+
+        int N = 2048;
+        auto window = createBlackmanWindow(N);
+        std::vector<double> windowedSamples(N);
+
+        QVector<int> waveData;
+
+        for (int i = 0; i < N; ++i) {
+            waveData.push_back(static_cast<int>(std::clamp(static_cast<int>(128 * rawSamples[i] + 1) * 2, 0, 255)));
+            windowedSamples[i] = rawSamples[i] * window[i];
+        }
+
+        // Step 2: Convert to complex
+        std::vector<std::complex<double>> complexSamples(N);
+        for (int i = 0; i < N; ++i) {
+            complexSamples[i] = std::complex<double>(windowedSamples[i], 0.0);
+        }
+
+        // Step 3: Apply FFTW3 transformation
+        fftw_plan plan = fftw_plan_dft_1d(N,
+                                          reinterpret_cast<fftw_complex*>(complexSamples.data()),
+                                          reinterpret_cast<fftw_complex*>(complexSamples.data()),
+                                          FFTW_FORWARD, FFTW_ESTIMATE);
+
+        fftw_execute(plan);
+        fftw_destroy_plan(plan);
+
+        // Step 4: Convert back to floats and divide by N
+        std::vector<float> magnitude(N);
+        for (int i = 0; i < N; ++i) {
+            double real = complexSamples[i].real();
+            double imag = complexSamples[i].imag();
+            magnitude[i] = static_cast<float>(std::sqrt(real * real + imag * imag) / N);
+        }
+
+        // Step 5: Apply smoothing
+        auto smoothed = smoothData(magnitude, 3); // Using window size of 3
+
+        // Step 6: Convert to decibels
+        std::vector<float> dbValues(smoothed.size());
+        const float minDb = -100.0f; // Minimum dB value for clamping
+        const float reference = 1.0f; // Reference amplitude
+
+        for (size_t i = 0; i < smoothed.size(); ++i) {
+            if (smoothed[i] > 0) {
+                dbValues[i] = 20.0f * std::log10(smoothed[i] / reference);
+            } else {
+                dbValues[i] = minDb;
+            }
+        }
+
+        // Step 7: Clamp to 8-bit values for red channel
+        std::vector<uint8_t> redChannel(dbValues.size());
+
+        for (size_t i = 0; i < dbValues.size(); ++i) {
+            // Clamp between -100dB and 0dB, then map to 0-255 range
+            float clamped = std::max(minDb, std::min(0.0f, dbValues[i]));
+            redChannel[i] = static_cast<uint8_t>((clamped + 100.0f) * 2.55f);
+        }
+
+        QPixmap audioTexture(512,2);
+        QPainter painter(&audioTexture);
+        painter.fillRect(QRect(0,0,512,2), QColor::fromRgb(0,0,0));
+
+        //we can only paint the lower half of the spectrum
+        for(int index = 0; index < 512; ++index)
+        {
+            //paint the pixels
+            painter.setPen(QPen(QColor::fromRgb(redChannel[index], 0, 0), 1));
+            painter.drawPoint(index, 0);
+            painter.setPen(QPen(QColor::fromRgb(waveData[index], 0, 0), 1));
+            painter.drawPoint(index, 1);
+        }
+
+        painter.end();
+
+        if(m_mutex.tryLock(1))
+        {
+            m_instance->m_frame = audioTexture;
+            m_mutex.unlock();
+        }
+    }
+
+    pw_stream_queue_buffer(data->stream, b);
+}
+
+// Blackman window function
+std::vector<double> AudioModel::createBlackmanWindow(int size) {
+    std::vector<double> window(size);
+    const double a0 = 0.42;
+    const double a1 = 0.5;
+    const double a2 = 0.08;
+
+    for (int i = 0; i < size; ++i) {
+        window[i] = a0 - a1 * std::cos(2.0 * M_PI * i / (size - 1)) +
+                    a2 * std::cos(4.0 * M_PI * i / (size - 1));
+    }
+    return window;
+}
+
+// Simple smoothing function using moving average
+std::vector<float> AudioModel::smoothData(const std::vector<float>& data, int windowSize) {
+    std::vector<float> smoothed(data.size());
+
+    for (size_t i = 0; i < data.size(); ++i) {
+        float sum = 0.0f;
+        int count = 0;
+
+        for (int j = -windowSize/2; j <= windowSize/2; ++j) {
+            int idx = i + j;
+            if (idx >= 0 && idx < static_cast<int>(data.size())) {
+                sum += data[idx];
+                count++;
+            }
+        }
+
+        smoothed[i] = count > 0 ? sum / count : 0.0f;
+    }
+
+    return smoothed;
+}
+
+void AudioModel::do_quit(void *userdata, int signal_number)
+{
+    Q_UNUSED(signal_number)
+
+    struct impl *data = reinterpret_cast<impl*>(userdata);
+    pw_main_loop_quit(data->loop);
+}

plugin/AudioModel.h

@@ -3,6 +3,18 @@
  *  Copyright (C) 2025 @DigitalArtifex | github.com/DigitalArtifex
  *
  *  AudioModel.h
+ * 
+ *  This is pretty much just a reimplementation of the audiocapture example
+ *  from the PipeWire docs.
+ * 
+ *  NOTICE:
+ *  The spectrum data is currently out of spec according to the documentation
+ *  https://webaudio.github.io/web-audio-api/#smoothing-over-time
+ * 
+ *  The described smoothing method was resulting in inconsistent data. This
+ *  is likely to a poor implementation. A linear smoothing algo seems to work
+ *  (at least visually). Will need to revisit the temporal implementation if 
+ *  things do not work as expected.
  *
  *  This program is free software: you can redistribute it and/or modify
  *  it under the terms of the GNU General Public License as published by
@@ -22,85 +34,102 @@
 #define AUDIOMODEL_H
 #include "Komplex_global.h"
 
-#include <QObject>
+#include <QObject> 
 #include <QString>
 #include <QFile>
 #include <QJsonDocument>
 #include <QJsonObject>
 #include <QJsonArray>
 #include <QJsonParseError>
-#include <QAudioDevice>
+#include <QThread>
-#include <QMediaDevices>
+#include <QtEndian>
-#include <QAudioInput>
+#include <QPixmap>
-#include <QMediaCaptureSession>
+#include <QQmlEngine>
-#include <QMediaRecorder>
+#include <QJSValue>
+#include <QVector>
+#include <QPainter>
+#include <QBrush>
+#include <QPen>
+#include <QThread>
+#include <QMutex>
+#include <QtConcurrent/QtConcurrent>
 #include <QtQml/qqmlregistration.h>
 
+#include <complex>
+#include <pipewire/pipewire.h>
+#include <spa/param/audio/raw.h>
+#include <spa/pod/pod.h>
+#include <spa/pod/builder.h>
+#include <spa/param/format-types.h>
+#include <spa/param/buffers.h>
+#include <spa/param/audio/format-utils.h>
 
-    class KOMPLEX_EXPORT AudioModel : public QObject
+class KOMPLEX_EXPORT AudioModel : public QObject
+{
+    Q_OBJECT
+    QML_SINGLETON
+    QML_NAMED_ELEMENT(AudioModel)
+
+public:
+    AudioModel(QObject *parent = nullptr);
+    ~AudioModel();
+
+    /**!
+     * @brief frame
+     * This function returns the current audio frame as a QPixmap.
+     * It is expected to be called after the frameChanged signal is emitted, if using from CPP
+     *
+     * If it is being used from QML, it will need to be resolved from the AuidoTexture Image Provider (image:/audio/frame#.jpg).
+     * See AudioImage provider for more details.
+     * 
+     * @return QPixmap containing the current audio frame.
+     */
+    static QPixmap frame();
+
+    // Q_INVOKABLE bool init();
+    Q_INVOKABLE static void startCapture();
+    Q_INVOKABLE static void stopCapture();
+
+private Q_SLOTS:
+    static void startCaptureAsync();
+
+private:
+    static std::vector<double> createBlackmanWindow(int size);
+    static std::vector<float> smoothData(const std::vector<float>& data, int windowSize = 5);
+
+    struct impl
     {
-        Q_OBJECT
+        pw_main_loop *loop;
-        QML_ELEMENT
+        pw_stream *stream;
-    public:
-        explicit AudioModel(QObject *parent = nullptr);
-        ~AudioModel();
 
-        /**!
+        spa_audio_info format;
-         * @brief frame
+        unsigned move:1;
-         * This function returns the current audio frame as a QString.
-         * It is expected to be called periodically to update the audio frame for the shader.
-         * 
-         * @return QString containing the current audio frame.
-         */
-        QByteArray frame() const;
 
-        /**!
+        QVector<qreal> samples; // we need at least 2048 samples
-         * @brief device
+        QVector<qreal> smoothed; // we're supposed to save for smoothing, but I couldn't get this method to work
-         * This function returns the currently set audio device name.
+        qreal last;
-         * 
-         * @return QString containing the name of the audio device.
-         */
-        QString device() const;
-
-        /**!
-         * @brief availableDevices
-         * This function returns a list of available audio devices on the system.
-         * 
-         * @return QStringList containing the names of available audio devices.
-         */
-        QStringList availableDevices() const;
-
-        /**!
-         * @brief setDeviceName
-         * This function sets the audio device to be used for capturing audio frames.
-         * 
-         * @param device The name of the audio device to set.
-         */
-        Q_INVOKABLE void setDeviceName(const QString &device);
-
-        /**!
-         * @brief getAudioFrame
-         * This function retrieves the current audio frame from the specified audio device.
-         * It is expected to be called periodically to update the audio frame for the shader.
-         * 
-         * It is an asynchronous fuction and will emit the frameChanged signal when the audio frame is ready.
-         */
-        Q_INVOKABLE void getAudioFrame();
-
-    Q_SIGNALS:
-        void frameChanged();
-
-    private:
-        QString m_deviceString;
-
-        QMediaCaptureSession *m_captureSession = nullptr;
-        QAudioInput *m_audioInput = nullptr;
-        QMediaRecorder *m_recorder = nullptr;
-
-        Q_PROPERTY(QByteArray frame READ frame NOTIFY frameChanged)
-        Q_PROPERTY(QString device READ device WRITE setDeviceName NOTIFY frameChanged)
     };
 
+    inline static AudioModel *m_instance = nullptr;
+    inline static QThread *m_thread = nullptr;
+    inline static QMutex m_mutex;
+
+    QPixmap m_frame;
+
+    inline static impl m_impl_data;
+    inline static bool m_running = false;
+
+    static void on_process(void *user_data);
+    static void do_quit(void *user_data, int signal_number);
+
+    static void on_stream_param_changed(void *_data, uint32_t id, const struct spa_pod *param);
+
+    inline static const struct pw_stream_events stream_events = {
+        .version = PW_VERSION_STREAM_EVENTS,
+        .param_changed = on_stream_param_changed,
+        .process = on_process,
+    };
+};
 
 Q_DECLARE_METATYPE(AudioModel)
 

plugin/CMakeLists.txt

@@ -12,6 +12,8 @@ add_library(
         plugin.cpp
         ShaderPackModel.cpp
         AudioModel.cpp
+        AudioImageProvider.cpp
+        AudioImageProvider.h
 ) 
 
 qt_add_qml_module(
@@ -20,7 +22,7 @@ qt_add_qml_module(
         ${QMLPLUGIN_URI}
     VERSION
         1.0
-    PLUGIN_TARGET  
+    PLUGIN_TARGET
         ${PROJECT_NAME}
     CLASS_NAME
         KomplexPlugin
@@ -28,7 +30,8 @@ qt_add_qml_module(
         plugin.cpp
         ShaderPackModel.cpp
         AudioModel.cpp
-    NO_GENERATE_PLUGIN_SOURCE 
+        AudioImageProvider.cpp
+    NO_GENERATE_PLUGIN_SOURCE
 )
 
 target_link_libraries(
@@ -43,6 +46,8 @@ target_link_libraries(
         KF6::CoreAddons
         KF6::I18n
         KF6::Package
+        PipeWire::PipeWire
+        fftw3
 )
 
 target_compile_definitions(

plugin/plugin.cpp

@@ -3,6 +3,7 @@
 #include <QQmlExtensionPlugin>
 
 #include "AudioModel.h"
+#include "AudioImageProvider.h"
 #include "ShaderPackModel.h"
 #include "Komplex_global.h"
 
@@ -18,6 +19,12 @@ public:
         qmlRegisterType<AudioModel>(uri, 1, 0, "AudioModel");
         qmlRegisterType<ShaderPackModel>(uri, 1, 0, "ShaderPackModel");
     }
+
+    void initializeEngine(QQmlEngine *engine, const char *uri) override
+    {
+        Q_ASSERT(QLatin1String(uri) == QLatin1String("com.github.digitalartifex.komplex"));
+        engine->addImageProvider(QString::fromLatin1("audiotexture"), new AudioImageProvider);
+    }
 };
 
 #include "plugin.moc"