mirror of
https://github.com/RGBCube/serenity
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d049626f40
All audio applications (aplay, Piano, Sound Player) respect the ability of the system to have theoretically any sample rate. Therefore, they resample their own audio into the system sample rate. LibAudio previously had its loaders resample their own audio, even though they expose their sample rate. This is now changed. The loaders output audio data in their file's sample rate, which the user has to query and resample appropriately. Resampling code from Buffer, WavLoader and FlacLoader is removed. Note that these applications only check the sample rate at startup, which is reasonable (the user has to restart applications when changing the sample rate). Fully dynamic adaptation could both lead to errors and will require another IPC interface. This seems to be enough for now.
63 lines
1.9 KiB
C++
63 lines
1.9 KiB
C++
/*
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* Copyright (c) 2018-2020, Andreas Kling <kling@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/Vector.h>
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#include <LibAudio/Buffer.h>
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#include <LibAudio/ClientConnection.h>
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#include <LibAudio/Loader.h>
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#include <LibCore/Timer.h>
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class PlaybackManager final {
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public:
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PlaybackManager(NonnullRefPtr<Audio::ClientConnection>);
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~PlaybackManager();
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void play();
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void stop();
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void pause();
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void seek(const int position);
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void loop(bool);
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bool toggle_pause();
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void set_loader(NonnullRefPtr<Audio::Loader>&&);
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size_t device_sample_rate() const { return m_device_sample_rate; }
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int last_seek() const { return m_last_seek; }
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bool is_paused() const { return m_paused; }
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float total_length() const { return m_total_length; }
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RefPtr<Audio::Buffer> current_buffer() const { return m_current_buffer; }
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NonnullRefPtr<Audio::ClientConnection> connection() const { return m_connection; }
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Function<void()> on_update;
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Function<void(Audio::Buffer&)> on_load_sample_buffer;
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Function<void()> on_finished_playing;
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private:
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void next_buffer();
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void set_paused(bool);
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bool m_paused { true };
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bool m_loop = { false };
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size_t m_last_seek { 0 };
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float m_total_length { 0 };
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// FIXME: Get this from the audio server
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size_t m_device_sample_rate { 44100 };
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size_t m_device_samples_per_buffer { 0 };
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size_t m_source_buffer_size_bytes { 0 };
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RefPtr<Audio::Loader> m_loader { nullptr };
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NonnullRefPtr<Audio::ClientConnection> m_connection;
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RefPtr<Audio::Buffer> m_current_buffer;
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Optional<Audio::ResampleHelper<double>> m_resampler;
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RefPtr<Core::Timer> m_timer;
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// Controls the GUI update rate. A smaller value makes the visualizations nicer.
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static constexpr u32 update_rate_ms = 50;
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// Number of milliseconds of audio data contained in each audio buffer
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static constexpr u32 buffer_size_ms = 100;
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};
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