AudioServer: Move ClientAudioStream to own files

This class will only grow, and it should really have its own files.

Userland/Services/AudioServer/ClientAudioStream.cpp

@@ -0,0 +1,121 @@
+/*
+ * Copyright (c) 2018-2022, the SerenityOS developers.
+ * Copyright (c) 2021-2023, kleines Filmröllchen <filmroellchen@serenityos.org>
+ *
+ * SPDX-License-Identifier: BSD-2-Clause
+ */
+
+#include "ClientAudioStream.h"
+#include <LibAudio/Resampler.h>
+
+namespace AudioServer {
+
+ClientAudioStream::ClientAudioStream(ConnectionFromClient& client)
+    : m_client(client)
+{
+}
+
+ConnectionFromClient* ClientAudioStream::client()
+{
+    return m_client.ptr();
+}
+
+bool ClientAudioStream::is_connected() const
+{
+    return m_client && m_client->is_open();
+}
+
+bool ClientAudioStream::get_next_sample(Audio::Sample& sample, u32 audiodevice_sample_rate)
+{
+    // Note: Even though we only check client state here, we will probably close the client much earlier.
+    if (!is_connected())
+        return false;
+
+    if (m_paused)
+        return false;
+
+    if (m_in_chunk_location >= m_current_audio_chunk.size()) {
+        auto result = m_buffer->dequeue();
+        if (result.is_error()) {
+            if (result.error() == Audio::AudioQueue::QueueStatus::Empty) {
+                dbgln_if(AUDIO_DEBUG, "Audio client {} can't keep up!", m_client->client_id());
+            }
+
+            return false;
+        }
+        // FIXME: Our resampler and the way we resample here are bad.
+        //        Ideally, we should both do perfect band-corrected resampling,
+        //        as well as carry resampling state over between buffers.
+        auto attempted_resample = Audio::ResampleHelper<Audio::Sample> {
+            m_sample_rate == 0 ? audiodevice_sample_rate : m_sample_rate, audiodevice_sample_rate
+        }
+                                      .try_resample(result.release_value());
+        if (attempted_resample.is_error())
+            return false;
+
+        // If the sample rate changes underneath us, we will still play the existing buffer unchanged until we're done.
+        // This is not a significant problem since the buffers are very small (~100 samples or less).
+        m_current_audio_chunk = attempted_resample.release_value();
+        m_in_chunk_location = 0;
+    }
+
+    sample = m_current_audio_chunk[m_in_chunk_location++];
+
+    return true;
+}
+
+void ClientAudioStream::set_buffer(OwnPtr<Audio::AudioQueue> buffer)
+{
+    m_buffer = move(buffer);
+}
+
+void ClientAudioStream::clear()
+{
+    ErrorOr<Array<Audio::Sample, Audio::AUDIO_BUFFER_SIZE>, Audio::AudioQueue::QueueStatus> result = Audio::AudioQueue::QueueStatus::Invalid;
+    do {
+        result = m_buffer->dequeue();
+    } while (!result.is_error() || result.error() != Audio::AudioQueue::QueueStatus::Empty);
+}
+
+void ClientAudioStream::set_paused(bool paused)
+{
+    m_paused = paused;
+}
+
+FadingProperty<double>& ClientAudioStream::volume()
+{
+    return m_volume;
+}
+
+double ClientAudioStream::volume() const
+{
+    return m_volume;
+}
+
+void ClientAudioStream::set_volume(double const volume)
+{
+    m_volume = volume;
+}
+
+bool ClientAudioStream::is_muted() const
+{
+    return m_muted;
+}
+
+void ClientAudioStream::set_muted(bool muted)
+{
+    m_muted = muted;
+}
+
+u32 ClientAudioStream::sample_rate() const
+{
+    return m_sample_rate;
+}
+
+void ClientAudioStream::set_sample_rate(u32 sample_rate)
+{
+    dbgln_if(AUDIO_DEBUG, "queue {} got sample rate {} Hz", m_client->client_id(), sample_rate);
+    m_sample_rate = sample_rate;
+}
+
+}