text: /*
   * 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 ConnectionFromClient final
      : public IPC::ConnectionToServer<AudioClientEndpoint, AudioServerEndpoint>
      , public AudioClientEndpoint {
      IPC_CLIENT_CONNECTION(ConnectionFromClient, "/tmp/portal/audio")
  public:
      virtual ~ConnectionFromClient() 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:
      ConnectionFromClient(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 {};
  };
  
  }
  