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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.
127 lines
3.1 KiB
C++
127 lines
3.1 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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#include "PlaybackManager.h"
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PlaybackManager::PlaybackManager(NonnullRefPtr<Audio::ClientConnection> connection)
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: m_connection(connection)
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{
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m_timer = Core::Timer::construct(PlaybackManager::update_rate_ms, [&]() {
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if (!m_loader)
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return;
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next_buffer();
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});
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m_timer->stop();
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m_device_sample_rate = connection->get_sample_rate();
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}
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PlaybackManager::~PlaybackManager()
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{
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}
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void PlaybackManager::set_loader(NonnullRefPtr<Audio::Loader>&& loader)
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{
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stop();
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m_loader = loader;
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if (m_loader) {
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m_total_length = m_loader->total_samples() / static_cast<float>(m_loader->sample_rate());
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m_device_samples_per_buffer = PlaybackManager::buffer_size_ms / 1000.0f * m_device_sample_rate;
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u32 source_samples_per_buffer = PlaybackManager::buffer_size_ms / 1000.0f * m_loader->sample_rate();
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m_source_buffer_size_bytes = source_samples_per_buffer * m_loader->num_channels() * m_loader->bits_per_sample() / 8;
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m_resampler = Audio::ResampleHelper<double>(m_loader->sample_rate(), m_device_sample_rate);
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m_timer->start();
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} else {
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m_timer->stop();
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}
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}
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void PlaybackManager::stop()
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{
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set_paused(true);
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m_connection->clear_buffer(true);
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m_last_seek = 0;
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m_current_buffer = nullptr;
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if (m_loader)
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m_loader->reset();
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}
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void PlaybackManager::play()
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{
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set_paused(false);
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}
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void PlaybackManager::loop(bool loop)
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{
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m_loop = loop;
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}
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void PlaybackManager::seek(const int position)
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{
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if (!m_loader)
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return;
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m_last_seek = position;
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bool paused_state = m_paused;
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set_paused(true);
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m_connection->clear_buffer(true);
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m_current_buffer = nullptr;
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m_loader->seek(position);
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if (!paused_state)
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set_paused(false);
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}
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void PlaybackManager::pause()
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{
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set_paused(true);
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}
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void PlaybackManager::set_paused(bool paused)
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{
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m_paused = paused;
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m_connection->set_paused(paused);
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}
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bool PlaybackManager::toggle_pause()
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{
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if (m_paused) {
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play();
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} else {
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pause();
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}
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return m_paused;
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}
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void PlaybackManager::next_buffer()
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{
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if (on_update)
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on_update();
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if (m_paused)
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return;
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u32 audio_server_remaining_samples = m_connection->get_remaining_samples();
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bool all_samples_loaded = (m_loader->loaded_samples() >= m_loader->total_samples());
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bool audio_server_done = (audio_server_remaining_samples == 0);
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if (all_samples_loaded && audio_server_done) {
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stop();
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if (on_finished_playing)
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on_finished_playing();
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return;
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}
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if (audio_server_remaining_samples < m_device_samples_per_buffer) {
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m_current_buffer = m_loader->get_more_samples(m_source_buffer_size_bytes);
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VERIFY(m_resampler.has_value());
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m_resampler->reset();
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m_current_buffer = Audio::resample_buffer(m_resampler.value(), *m_current_buffer);
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if (m_current_buffer)
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m_connection->enqueue(*m_current_buffer);
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}
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}
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