serenity/Userland/Services/AudioServer/Mixer.h

/*
 * Copyright (c) 2018-2020, Andreas Kling <kling@serenityos.org>
 * Copyright (c) 2021-2022, kleines Filmröllchen <filmroellchen@serenityos.org>
 *
 * SPDX-License-Identifier: BSD-2-Clause
 */

#pragma once

#include "ConnectionFromClient.h"
#include "FadingProperty.h"
#include <AK/Atomic.h>
#include <AK/Badge.h>
#include <AK/ByteBuffer.h>
#include <AK/Debug.h>
#include <AK/Queue.h>
#include <AK/RefCounted.h>
#include <AK/WeakPtr.h>
#include <LibAudio/Queue.h>
#include <LibAudio/Resampler.h>
#include <LibCore/ConfigFile.h>
#include <LibCore/File.h>
#include <LibCore/Timer.h>
#include <LibThreading/ConditionVariable.h>
#include <LibThreading/Mutex.h>
#include <LibThreading/Thread.h>
#include <sys/types.h>

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 = 0.95;
// The size of the buffer in samples that the hardware receives through write() calls to the audio device.
constexpr size_t HARDWARE_BUFFER_SIZE = 512;
// The hardware buffer size in bytes; there's two channels of 16-bit samples.
constexpr size_t HARDWARE_BUFFER_SIZE_BYTES = HARDWARE_BUFFER_SIZE * 2 * sizeof(i16);

class ConnectionFromClient;

class ClientAudioStream : public RefCounted<ClientAudioStream> {
public:
    explicit ClientAudioStream(ConnectionFromClient&);
    ~ClientAudioStream() = default;

    bool get_next_sample(Audio::Sample& sample, u32 audiodevice_sample_rate)
    {
        // Note: Even though we only check client state here, we will probably close the client much earlier.
        if (!is_connected())
            return false;

        if (m_paused)
            return false;

        if (m_in_chunk_location >= m_current_audio_chunk.size()) {
            auto result = m_buffer->dequeue();
            if (result.is_error()) {
                if (result.error() == Audio::AudioQueue::QueueStatus::Empty) {
                    dbgln_if(AUDIO_DEBUG, "Audio client {} can't keep up!", m_client->client_id());
                }

                return false;
            }
            // FIXME: Our resampler and the way we resample here are bad.
            //        Ideally, we should both do perfect band-corrected resampling,
            //        as well as carry resampling state over between buffers.
            auto attempted_resample = Audio::ResampleHelper<Audio::Sample> {
                m_sample_rate == 0 ? audiodevice_sample_rate : m_sample_rate, audiodevice_sample_rate
            }
                                          .try_resample(result.release_value());
            if (attempted_resample.is_error())
                return false;

            // If the sample rate changes underneath us, we will still play the existing buffer unchanged until we're done.
            // This is not a significant problem since the buffers are very small (~100 samples or less).
            m_current_audio_chunk = attempted_resample.release_value();
            m_in_chunk_location = 0;
        }

        sample = m_current_audio_chunk[m_in_chunk_location++];

        return true;
    }

    bool is_connected() const { return m_client && m_client->is_open(); }

    ConnectionFromClient* client() { return m_client.ptr(); }

    void set_buffer(OwnPtr<Audio::AudioQueue> buffer) { m_buffer = move(buffer); }

    void clear()
    {
        ErrorOr<Array<Audio::Sample, Audio::AUDIO_BUFFER_SIZE>, Audio::AudioQueue::QueueStatus> result = Audio::AudioQueue::QueueStatus::Invalid;
        do {
            result = m_buffer->dequeue();
        } while (!result.is_error() || result.error() != Audio::AudioQueue::QueueStatus::Empty);
    }

    void set_paused(bool paused) { m_paused = paused; }

    FadingProperty<double>& volume() { return m_volume; }
    double volume() const { return m_volume; }
    void set_volume(double const volume) { m_volume = volume; }
    bool is_muted() const { return m_muted; }
    void set_muted(bool muted) { m_muted = muted; }
    u32 sample_rate() const { return m_sample_rate; }
    void set_sample_rate(u32 sample_rate)
    {
        dbgln_if(AUDIO_DEBUG, "queue {} got sample rate {} Hz", m_client->client_id(), sample_rate);
        m_sample_rate = sample_rate;
    }

private:
    OwnPtr<Audio::AudioQueue> m_buffer;
    Vector<Audio::Sample> m_current_audio_chunk;
    size_t m_in_chunk_location;

    bool m_paused { true };
    bool m_muted { false };
    u32 m_sample_rate;

    WeakPtr<ConnectionFromClient> m_client;
    FadingProperty<double> m_volume { 1 };
};

class Mixer : public Core::Object {
    C_OBJECT_ABSTRACT(Mixer)
public:
    static ErrorOr<NonnullRefPtr<Mixer>> try_create(NonnullRefPtr<Core::ConfigFile> config)
    {
        // FIXME: Allow AudioServer to use other audio channels as well
        auto device = TRY(Core::File::open("/dev/audio/0"sv, Core::File::OpenMode::Write));
        return adopt_nonnull_ref_or_enomem(new (nothrow) Mixer(move(config), move(device)));
    }

    virtual ~Mixer() override = default;

    NonnullRefPtr<ClientAudioStream> create_queue(ConnectionFromClient&);

    // 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);

    int audiodevice_set_sample_rate(u32 sample_rate);
    u32 audiodevice_get_sample_rate() const;

private:
    Mixer(NonnullRefPtr<Core::ConfigFile> config, NonnullOwnPtr<Core::File> device);

    void request_setting_sync();

    Vector<NonnullRefPtr<ClientAudioStream>> m_pending_mixing;
    Threading::Mutex m_pending_mutex;
    Threading::ConditionVariable m_mixing_necessary { m_pending_mutex };

    NonnullOwnPtr<Core::File> m_device;
    mutable Optional<u32> m_cached_sample_rate {};

    NonnullRefPtr<Threading::Thread> m_sound_thread;

    bool m_muted { false };
    FadingProperty<double> m_main_volume { 1 };

    NonnullRefPtr<Core::ConfigFile> m_config;
    RefPtr<Core::Timer> m_config_write_timer;

    Array<u8, HARDWARE_BUFFER_SIZE_BYTES> m_stream_buffer;
    Array<u8, HARDWARE_BUFFER_SIZE_BYTES> const m_zero_filled_buffer {};

    void mix();
};

// Interval in ms when the server tries to save its configuration to disk.
constexpr unsigned AUDIO_CONFIG_WRITE_INTERVAL = 2000;

}