mirror of
https://github.com/RGBCube/serenity
synced 2025-10-31 17:22:43 +00:00
d905498fb6
- Use Optional references instead of pointers - Clean up some const and nullability weirdness - Use proper error return value for get_next_sample
117 lines
3.3 KiB
C++
117 lines
3.3 KiB
C++
/*
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* Copyright (c) 2018-2022, the SerenityOS developers.
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* Copyright (c) 2021-2023, 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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#include "ClientAudioStream.h"
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#include <LibAudio/Resampler.h>
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namespace AudioServer {
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ClientAudioStream::ClientAudioStream(ConnectionFromClient& client)
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: m_client(client)
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{
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}
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Optional<ConnectionFromClient&> ClientAudioStream::client()
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{
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return m_client.has_value() ? *m_client : Optional<ConnectionFromClient&> {};
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}
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bool ClientAudioStream::is_connected() const
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{
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return m_client && m_client->is_open();
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}
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ErrorOr<Audio::Sample, ClientAudioStream::ErrorState> ClientAudioStream::get_next_sample(u32 audiodevice_sample_rate)
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{
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// Note: Even though we only check client state here, we will probably close the client much earlier.
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if (!is_connected())
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return ErrorState::ClientDisconnected;
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if (m_paused)
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return ErrorState::ClientUnderrun;
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if (m_in_chunk_location >= m_current_audio_chunk.size()) {
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auto result = m_buffer->dequeue();
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if (result.is_error()) {
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if (result.error() == Audio::AudioQueue::QueueStatus::Empty) {
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dbgln_if(AUDIO_DEBUG, "Audio client {} can't keep up!", m_client->client_id());
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}
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return ErrorState::ClientUnderrun;
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}
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// FIXME: Our resampler and the way we resample here are bad.
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// Ideally, we should both do perfect band-corrected resampling,
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// as well as carry resampling state over between buffers.
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auto maybe_resampled = Audio::ResampleHelper<Audio::Sample> { m_sample_rate == 0 ? audiodevice_sample_rate : m_sample_rate, audiodevice_sample_rate }
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.try_resample(result.release_value());
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if (maybe_resampled.is_error())
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return ErrorState::ResamplingError;
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// If the sample rate changes underneath us, we will still play the existing buffer unchanged until we're done.
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// This is not a significant problem since the buffers are very small (~100 samples or less).
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m_current_audio_chunk = maybe_resampled.release_value();
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m_in_chunk_location = 0;
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}
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return m_current_audio_chunk[m_in_chunk_location++];
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}
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void ClientAudioStream::set_buffer(NonnullOwnPtr<Audio::AudioQueue> buffer)
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{
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m_buffer = move(buffer);
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}
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void ClientAudioStream::clear()
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{
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ErrorOr<Array<Audio::Sample, Audio::AUDIO_BUFFER_SIZE>, Audio::AudioQueue::QueueStatus> result = Audio::AudioQueue::QueueStatus::Invalid;
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do {
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result = m_buffer->dequeue();
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} while (!result.is_error() || result.error() != Audio::AudioQueue::QueueStatus::Empty);
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}
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void ClientAudioStream::set_paused(bool paused)
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{
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m_paused = paused;
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}
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FadingProperty<double>& ClientAudioStream::volume()
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{
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return m_volume;
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}
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double ClientAudioStream::volume() const
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{
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return m_volume;
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}
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void ClientAudioStream::set_volume(double const volume)
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{
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m_volume = volume;
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}
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bool ClientAudioStream::is_muted() const
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{
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return m_muted;
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}
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void ClientAudioStream::set_muted(bool muted)
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{
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m_muted = muted;
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}
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u32 ClientAudioStream::sample_rate() const
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{
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return m_sample_rate;
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}
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void ClientAudioStream::set_sample_rate(u32 sample_rate)
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{
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dbgln_if(AUDIO_DEBUG, "queue {} got sample rate {} Hz", m_client->client_id(), sample_rate);
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m_sample_rate = sample_rate;
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}
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}
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