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https://github.com/RGBCube/serenity
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AudioServer: Move ClientAudioStream to own files
This class will only grow, and it should really have its own files.
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6 changed files with 181 additions and 95 deletions
121
Userland/Services/AudioServer/ClientAudioStream.cpp
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121
Userland/Services/AudioServer/ClientAudioStream.cpp
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/*
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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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ConnectionFromClient* ClientAudioStream::client()
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{
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return m_client.ptr();
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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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bool ClientAudioStream::get_next_sample(Audio::Sample& 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 false;
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if (m_paused)
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return false;
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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 false;
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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 attempted_resample = Audio::ResampleHelper<Audio::Sample> {
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m_sample_rate == 0 ? audiodevice_sample_rate : m_sample_rate, audiodevice_sample_rate
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}
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.try_resample(result.release_value());
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if (attempted_resample.is_error())
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return false;
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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 = attempted_resample.release_value();
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m_in_chunk_location = 0;
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}
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sample = m_current_audio_chunk[m_in_chunk_location++];
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return true;
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}
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void ClientAudioStream::set_buffer(OwnPtr<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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