mirror of
https://github.com/LadybirdBrowser/ladybird.git
synced 2024-12-29 14:14:45 +03:00
3123753e6b
This is a temporary fix until we move AudioPlayerLoop to direct buffer enqueuing.
74 lines
2.5 KiB
C++
74 lines
2.5 KiB
C++
/*
|
|
* Copyright (c) 2021, kleines Filmröllchen <filmroellchen@serenityos.org>
|
|
* Copyright (c) 2021, JJ Roberts-White <computerfido@gmail.com>
|
|
*
|
|
* SPDX-License-Identifier: BSD-2-Clause
|
|
*/
|
|
|
|
#include "AudioPlayerLoop.h"
|
|
|
|
#include "TrackManager.h"
|
|
#include <AK/FixedArray.h>
|
|
#include <AK/NumericLimits.h>
|
|
#include <LibAudio/ConnectionToServer.h>
|
|
#include <LibAudio/Resampler.h>
|
|
#include <LibAudio/Sample.h>
|
|
#include <LibCore/EventLoop.h>
|
|
|
|
AudioPlayerLoop::AudioPlayerLoop(TrackManager& track_manager, bool& need_to_write_wav, Audio::WavWriter& wav_writer)
|
|
: m_track_manager(track_manager)
|
|
, m_buffer(FixedArray<DSP::Sample>::must_create_but_fixme_should_propagate_errors(sample_count))
|
|
, m_need_to_write_wav(need_to_write_wav)
|
|
, m_wav_writer(wav_writer)
|
|
{
|
|
m_audio_client = Audio::ConnectionToServer::try_create().release_value_but_fixme_should_propagate_errors();
|
|
|
|
auto target_sample_rate = m_audio_client->get_sample_rate();
|
|
if (target_sample_rate == 0)
|
|
target_sample_rate = Music::sample_rate;
|
|
m_resampler = Audio::ResampleHelper<DSP::Sample>(Music::sample_rate, target_sample_rate);
|
|
|
|
// FIXME: I said I would never write such a hack again, but here we are.
|
|
// This code should die as soon as possible anyways, so it doesn't matter.
|
|
// Please don't use this as an example to write good audio code; it's just here as a temporary hack.
|
|
Core::EventLoop::register_timer(*this, 5, true, Core::TimerShouldFireWhenNotVisible::Yes);
|
|
}
|
|
|
|
void AudioPlayerLoop::timer_event(Core::TimerEvent&)
|
|
{
|
|
while (m_audio_client->remaining_samples() < sample_count)
|
|
enqueue_audio();
|
|
}
|
|
|
|
void AudioPlayerLoop::enqueue_audio()
|
|
{
|
|
m_track_manager.fill_buffer(m_buffer);
|
|
// FIXME: Handle OOM better.
|
|
auto audio_buffer = m_resampler->resample(m_buffer);
|
|
(void)m_audio_client->async_enqueue(audio_buffer);
|
|
|
|
// FIXME: This should be done somewhere else.
|
|
if (m_need_to_write_wav) {
|
|
m_need_to_write_wav = false;
|
|
m_track_manager.reset();
|
|
m_track_manager.set_should_loop(false);
|
|
do {
|
|
m_track_manager.fill_buffer(m_buffer);
|
|
m_wav_writer.write_samples(m_buffer.span());
|
|
} while (m_track_manager.transport()->time());
|
|
m_track_manager.reset();
|
|
m_track_manager.set_should_loop(true);
|
|
m_wav_writer.finalize();
|
|
}
|
|
}
|
|
|
|
void AudioPlayerLoop::toggle_paused()
|
|
{
|
|
m_should_play_audio = !m_should_play_audio;
|
|
|
|
if (m_should_play_audio)
|
|
m_audio_client->async_start_playback();
|
|
else
|
|
m_audio_client->async_pause_playback();
|
|
}
|