mirror of
https://github.com/RGBCube/serenity
synced 2025-10-31 19:22:45 +00:00
93 lines
3.6 KiB
C++
93 lines
3.6 KiB
C++
/*
|
|
* 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/session/%sid/portal/audio"sv)
|
|
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);
|
|
ErrorOr<void> blocking_realtime_enqueue(Array<Sample, AUDIO_BUFFER_SIZE> samples, Function<void()> wait_function);
|
|
|
|
// 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 { nullptr };
|
|
Threading::Mutex m_enqueuer_loop_destruction;
|
|
|
|
// A good amount of time to sleep when the queue is full.
|
|
// (Only used for non-realtime enqueues)
|
|
timespec m_good_sleep_time {};
|
|
};
|
|
|
|
}
|