Audio: Change how volume works

Across the entire audio system, audio now works in 0-1 terms instead of
0-100 as before. Therefore, volume is now a double instead of an int.
The master volume of the AudioServer changes smoothly through a
FadingProperty, preventing clicks. Finally, volume computations are done
with logarithmic scaling, which is more natural for the human ear.

Note that this could be 4-5 different commits, but as they change each
other's code all the time, it makes no sense to split them up.

Userland/Services/AudioServer/AudioClient.ipc

@@ -4,5 +4,5 @@ endpoint AudioClient
 {
     finished_playing_buffer(i32 buffer_id) =|
     muted_state_changed(bool muted) =|
-    main_mix_volume_changed(i32 volume) =|
+    main_mix_volume_changed(double volume) =|
 }

Userland/Services/AudioServer/AudioServer.ipc

@@ -5,8 +5,8 @@ endpoint AudioServer
     // Mixer functions
     set_muted(bool muted) => ()
     get_muted() => (bool muted)
-    get_main_mix_volume() => (i32 volume)
-    set_main_mix_volume(i32 volume) => ()
+    get_main_mix_volume() => (double volume)
+    set_main_mix_volume(double volume) => ()
 
     // Audio device
     set_sample_rate(u16 sample_rate) => ()

Userland/Services/AudioServer/ClientConnection.cpp

@@ -48,7 +48,7 @@ void ClientConnection::did_change_muted_state(Badge<Mixer>, bool muted)
     async_muted_state_changed(muted);
 }
 
-void ClientConnection::did_change_main_mix_volume(Badge<Mixer>, int volume)
+void ClientConnection::did_change_main_mix_volume(Badge<Mixer>, double volume)
 {
     async_main_mix_volume_changed(volume);
 }
@@ -58,7 +58,7 @@ Messages::AudioServer::GetMainMixVolumeResponse ClientConnection::get_main_mix_v
     return m_mixer.main_volume();
 }
 
-void ClientConnection::set_main_mix_volume(i32 volume)
+void ClientConnection::set_main_mix_volume(double volume)
 {
     m_mixer.set_main_volume(volume);
 }

Userland/Services/AudioServer/ClientConnection.h

@@ -28,7 +28,7 @@ public:
 
     void did_finish_playing_buffer(Badge<BufferQueue>, int buffer_id);
     void did_change_muted_state(Badge<Mixer>, bool muted);
-    void did_change_main_mix_volume(Badge<Mixer>, int volume);
+    void did_change_main_mix_volume(Badge<Mixer>, double volume);
 
     virtual void die() override;
 
@@ -36,7 +36,7 @@ public:
 
 private:
     virtual Messages::AudioServer::GetMainMixVolumeResponse get_main_mix_volume() override;
-    virtual void set_main_mix_volume(i32) override;
+    virtual void set_main_mix_volume(double) override;
     virtual Messages::AudioServer::EnqueueBufferResponse enqueue_buffer(Core::AnonymousBuffer const&, i32, int) override;
     virtual Messages::AudioServer::GetRemainingSamplesResponse get_remaining_samples() override;
     virtual Messages::AudioServer::GetPlayedSamplesResponse get_played_samples() override;

Userland/Services/AudioServer/FadingProperty.h

@@ -0,0 +1,85 @@
+/*
+ * Copyright (c) 2021, kleines Filmröllchen <malu.bertsch@gmail.com>.
+ *
+ * SPDX-License-Identifier: BSD-2-Clause
+ */
+
+#pragma once
+
+#include "Mixer.h"
+#include <compare>
+
+namespace AudioServer {
+
+// This is in buffer counts.
+// As each buffer is approx 1/40 of a second, this means about 1/4 of a second of fade time.
+constexpr int DEFAULT_FADE_TIME = 10;
+
+// A property of an audio system that needs to fade briefly whenever changed.
+template<typename T>
+class FadingProperty {
+public:
+    FadingProperty(T const value)
+        : FadingProperty(value, DEFAULT_FADE_TIME)
+    {
+    }
+    FadingProperty(T const value, int const fade_time)
+        : m_old_value(value)
+        , m_new_value(move(value))
+        , m_fade_time(fade_time)
+    {
+    }
+    virtual ~FadingProperty()
+    {
+        m_old_value.~T();
+        m_new_value.~T();
+    }
+
+    FadingProperty<T>& operator=(T const& new_value)
+    {
+        // The origin of the fade is wherever we're right now.
+        m_old_value = static_cast<T>(*this);
+        m_new_value = new_value;
+        m_current_fade = 0;
+        return *this;
+    }
+    FadingProperty<T>& operator=(FadingProperty<T> const&) = delete;
+
+    operator T() const
+    {
+        if (!is_fading())
+            return m_new_value;
+        return m_old_value * (1 - m_current_fade) + m_new_value * (m_current_fade);
+    }
+
+    auto operator<=>(FadingProperty<T> const& other) const
+    {
+        return static_cast<T>(this) <=> static_cast<T>(other);
+    }
+
+    auto operator<=>(T const& other) const
+    {
+        return static_cast<T>(*this) <=> other;
+    }
+
+    void advance_time()
+    {
+        m_current_fade += 1.0 / static_cast<double>(m_fade_time);
+        m_current_fade = clamp(m_current_fade, 0.0, 1.0);
+    }
+
+    bool is_fading() const
+    {
+        return m_current_fade < 1;
+    }
+
+    T target() const { return m_new_value; }
+
+private:
+    T m_old_value {};
+    T m_new_value {};
+    double m_current_fade { 0 };
+    int const m_fade_time;
+};
+
+}

Userland/Services/AudioServer/Mixer.cpp

@@ -40,7 +40,7 @@ Mixer::Mixer(NonnullRefPtr<Core::ConfigFile> config)
     pthread_cond_init(&m_pending_cond, nullptr);
 
     m_muted = m_config->read_bool_entry("Master", "Mute", false);
-    m_main_volume = m_config->read_num_entry("Master", "Volume", 100);
+    m_main_volume = static_cast<double>(m_config->read_num_entry("Master", "Volume", 100)) / 100.0;
 
     m_sound_thread->start();
 }
@@ -78,18 +78,23 @@ void Mixer::mix()
         Audio::Frame mixed_buffer[1024];
         auto mixed_buffer_length = (int)(sizeof(mixed_buffer) / sizeof(Audio::Frame));
 
+        m_main_volume.advance_time();
+
+        int active_queues = 0;
         // Mix the buffers together into the output
         for (auto& queue : active_mix_queues) {
             if (!queue->client()) {
                 queue->clear();
                 continue;
             }
+            ++active_queues;
 
             for (int i = 0; i < mixed_buffer_length; ++i) {
                 auto& mixed_sample = mixed_buffer[i];
                 Audio::Frame sample;
                 if (!queue->get_next_sample(sample))
                     break;
+                sample.log_multiply(SAMPLE_HEADROOM);
                 mixed_sample += sample;
             }
         }
@@ -103,7 +108,11 @@ void Mixer::mix()
             for (int i = 0; i < mixed_buffer_length; ++i) {
                 auto& mixed_sample = mixed_buffer[i];
 
-                mixed_sample.scale(m_main_volume);
+                // Even though it's not realistic, the user expects no sound at 0%.
+                if (m_main_volume < 0.01)
+                    mixed_sample = { 0 };
+                else
+                    mixed_sample.log_multiply(m_main_volume);
                 mixed_sample.clip();
 
                 LittleEndian<i16> out_sample;
@@ -121,20 +130,20 @@ void Mixer::mix()
     }
 }
 
-void Mixer::set_main_volume(int volume)
+void Mixer::set_main_volume(double volume)
 {
     if (volume < 0)
         m_main_volume = 0;
-    else if (volume > 200)
-        m_main_volume = 200;
+    else if (volume > 2)
+        m_main_volume = 2;
     else
         m_main_volume = volume;
 
-    m_config->write_num_entry("Master", "Volume", volume);
+    m_config->write_num_entry("Master", "Volume", static_cast<int>(volume * 100));
     request_setting_sync();
 
     ClientConnection::for_each([&](ClientConnection& client) {
-        client.did_change_main_mix_volume({}, m_main_volume);
+        client.did_change_main_mix_volume({}, main_volume());
     });
 }
 

Userland/Services/AudioServer/Mixer.h

@@ -8,6 +8,7 @@
 #pragma once
 
 #include "ClientConnection.h"
+#include "FadingProperty.h"
 #include <AK/Atomic.h>
 #include <AK/Badge.h>
 #include <AK/ByteBuffer.h>
@@ -23,6 +24,10 @@
 
 namespace AudioServer {
 
+// Headroom, i.e. fixed attenuation for all audio streams.
+// This is to prevent clipping when two streams with low headroom (e.g. normalized & compressed) are playing.
+constexpr double SAMPLE_HEADROOM = 0.7;
+
 class ClientConnection;
 
 class BufferQueue : public RefCounted<BufferQueue> {
@@ -82,6 +87,10 @@ public:
         return -1;
     }
 
+    FadingProperty<double>& volume() { return m_volume; }
+    double volume() const { return m_volume; }
+    void set_volume(double const volume) { m_volume = volume; }
+
 private:
     RefPtr<Audio::Buffer> m_current;
     Queue<NonnullRefPtr<Audio::Buffer>> m_queue;
@@ -89,7 +98,9 @@ private:
     int m_remaining_samples { 0 };
     int m_played_samples { 0 };
     bool m_paused { false };
+
     WeakPtr<ClientConnection> m_client;
+    FadingProperty<double> m_volume { 1 };
 };
 
 class Mixer : public Core::Object {
@@ -100,8 +111,9 @@ public:
 
     NonnullRefPtr<BufferQueue> create_queue(ClientConnection&);
 
-    int main_volume() const { return m_main_volume; }
-    void set_main_volume(int volume);
+    // To the outside world, we pretend that the target volume is already reached, even though it may be still fading.
+    double main_volume() const { return m_main_volume.target(); }
+    void set_main_volume(double volume);
 
     bool is_muted() const { return m_muted; }
     void set_muted(bool);
@@ -122,7 +134,7 @@ private:
     NonnullRefPtr<Threading::Thread> m_sound_thread;
 
     bool m_muted { false };
-    int m_main_volume { 100 };
+    FadingProperty<double> m_main_volume { 1 };
 
     NonnullRefPtr<Core::ConfigFile> m_config;
     RefPtr<Core::Timer> m_config_write_timer;