LibAudio: Rename ConnectionFromClient to ConnectionToServer

The automatic nomenclature change for IPC sockets got this one wrong.

Userland/Libraries/LibAudio/ConnectionToServer.h

@@ -0,0 +1,93 @@
+/*
+ * Copyright (c) 2018-2020, Andreas Kling <kling@serenityos.org>
+ * Copyright (c) 2022, kleines Filmröllchen <filmroellchen@serenityos.org>
+ *
+ * SPDX-License-Identifier: BSD-2-Clause
+ */
+
+#pragma once
+
+#include <AK/Concepts.h>
+#include <AK/FixedArray.h>
+#include <AK/NonnullOwnPtr.h>
+#include <AK/OwnPtr.h>
+#include <LibAudio/Queue.h>
+#include <LibAudio/UserSampleQueue.h>
+#include <LibCore/EventLoop.h>
+#include <LibCore/Object.h>
+#include <LibIPC/ConnectionToServer.h>
+#include <LibThreading/Mutex.h>
+#include <LibThreading/Thread.h>
+#include <Userland/Services/AudioServer/AudioClientEndpoint.h>
+#include <Userland/Services/AudioServer/AudioServerEndpoint.h>
+
+namespace Audio {
+
+class ConnectionToServer final
+    : public IPC::ConnectionToServer<AudioClientEndpoint, AudioServerEndpoint>
+    , public AudioClientEndpoint {
+    IPC_CLIENT_CONNECTION(ConnectionToServer, "/tmp/portal/audio")
+public:
+    virtual ~ConnectionToServer() override;
+
+    // Both of these APIs are for convenience and when you don't care about real-time behavior.
+    // They will not work properly in conjunction with realtime_enqueue.
+    // If you don't refill the buffer in time with this API, the last shared buffer write is zero-padded to play all of the samples.
+    template<ArrayLike<Sample> Samples>
+    ErrorOr<void> async_enqueue(Samples&& samples)
+    {
+        return async_enqueue(TRY(FixedArray<Sample>::try_create(samples.span())));
+    }
+
+    ErrorOr<void> async_enqueue(FixedArray<Sample>&& samples);
+
+    void clear_client_buffer();
+
+    // Returns immediately with the appropriate status if the buffer is full; use in conjunction with remaining_buffers to get low latency.
+    ErrorOr<void, AudioQueue::QueueStatus> realtime_enqueue(Array<Sample, AUDIO_BUFFER_SIZE> samples);
+
+    // This information can be deducted from the shared audio buffer.
+    unsigned total_played_samples() const;
+    // How many samples remain in m_enqueued_samples.
+    unsigned remaining_samples();
+    // How many buffers (i.e. short sample arrays) the server hasn't played yet.
+    // Non-realtime code needn't worry about this.
+    size_t remaining_buffers() const;
+
+    virtual void die() override;
+
+    Function<void(bool muted)> on_main_mix_muted_state_change;
+    Function<void(double volume)> on_main_mix_volume_change;
+    Function<void(double volume)> on_client_volume_change;
+
+private:
+    ConnectionToServer(NonnullOwnPtr<Core::Stream::LocalSocket>);
+
+    virtual void main_mix_muted_state_changed(bool) override;
+    virtual void main_mix_volume_changed(double) override;
+    virtual void client_volume_changed(double) override;
+
+    // We use this to perform the audio enqueuing on the background thread's event loop
+    virtual void custom_event(Core::CustomEvent&) override;
+
+    // FIXME: This should be called every time the sample rate changes, but we just cautiously call it on every non-realtime enqueue.
+    void update_good_sleep_time();
+
+    // Shared audio buffer: both server and client constantly read and write to/from this.
+    // This needn't be mutex protected: it's internally multi-threading aware.
+    OwnPtr<AudioQueue> m_buffer;
+
+    // The queue of non-realtime audio provided by the user.
+    NonnullOwnPtr<UserSampleQueue> m_user_queue;
+
+    NonnullRefPtr<Threading::Thread> m_background_audio_enqueuer;
+    Core::EventLoop* m_enqueuer_loop;
+    Threading::Mutex m_enqueuer_loop_destruction;
+    Atomic<bool> m_audio_enqueuer_active { false };
+
+    // A good amount of time to sleep when the queue is full.
+    // (Only used for non-realtime enqueues)
+    timespec m_good_sleep_time {};
+};
+
+}