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/*
* 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);
// 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;
// A good amount of time to sleep when the queue is full.
// (Only used for non-realtime enqueues)
timespec m_good_sleep_time {};
};
}
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