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https://github.com/RGBCube/serenity
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125122a9ab
The audio enqueuer thread goes to sleep when there is no more audio data present, and through normal Core::EventLoop events it can be woken up. However, that waking up only happens when the thread is not currently running, so that the wake-up events don't queue up and cause weirdness. The atomic variable responsible for keeping track of whether the thread is active can lead to a racy deadlock however, where the audio enqueuer thread will never wake up again despite there being audio data to enqueue. Consider this scenario: - Main thread calls into async_enqueue. It detects that according to the atomic variable, the other thread is still running, skipping the event queue wake. - Enqueuer thread has just finished playing the last chunk of audio and detects that there is no audio left. It enters the if block with the dbgln "Reached end of provided audio data..." - Main thread enqueues audio, making the user sample queue non-empty. - Enqueuer thread does not check this condition again, instead setting the atomic variable to indicate that it is not running. It exits into an event loop sleep. - Main thread exits async_enqueue. The calling audio enqueuing system (see e.g. Piano, but all of them function similarly) will wait until the enqueuer thread has played enough samples before async_enqueue is called again. However, since the enqueuer thread will never play any audio, this condition is never fulfilled and audio playback deadlocks This commit fixes that by allowing the event loop to not enqueue an event that already exists, therefore overloading the audio enqueuer event loop by at maximum one message in weird situations. We entirely get rid of the atomic variable and the race condition is prevented.
92 lines
3.5 KiB
C++
92 lines
3.5 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/portal/audio")
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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>::try_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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// 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;
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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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