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