mirror of
https://github.com/LadybirdBrowser/ladybird.git
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9a2c80c791
This commit addresses two issues: 1. If you play a 96 KHz Wave file, the slider position is incorrect, because it is assumed all files are 44.1 KHz. 2. For high-bitrate files, there are audio dropouts due to not buffering enough audio data. Issue 1 is addressed by scaling the number of played samples by the ratio between the source and destination sample rates. Issue 2 is addressed by buffering a certain number of milliseconds worth of audio data (instead of a fixed number of bytes). This makes the the buffer size independent of the source sample rate. Some of the code is redesigned to be simpler. The code that did the book-keeping of which buffers need to be loaded and which have been already played has been removed. Instead, we enqueue a new buffer based on a low watermark of samples remaining in the audio server queue. Other small fixes include: 1. Disable the stop button when playback is finished. 2. Remove hard-coded instances of 44100. 3. Update the GUI every 50 ms (was 100), which improves visualizations.
122 lines
2.8 KiB
C++
122 lines
2.8 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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}
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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_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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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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