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https://github.com/SerenityOS/serenity.git
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b4fbd30b70
This change was a long time in the making ever since we obtained sample rate awareness in the system. Now, each client has its own sample rate, accessible via new IPC APIs, and the device sample rate is only accessible via the management interface. AudioServer takes care of resampling client streams into the device sample rate. Therefore, the main improvement introduced with this commit is full responsiveness to sample rate changes; all open audio programs will continue to play at correct speed with the audio resampled to the new device rate. The immediate benefits are manifold: - Gets rid of the legacy hardware sample rate IPC message in the non-managing client - Removes duplicate resampling and sample index rescaling code everywhere - Avoids potential sample index scaling bugs in SoundPlayer (which have happened many times before) and fixes a sample index scaling bug in aplay - Removes several FIXMEs - Reduces amount of sample copying in all applications (especially Piano, where this is critical), improving performance - Reduces number of resampling users, making future API changes (which will need to happen for correct resampling to be implemented) easier I also threw in a simple race condition fix for Piano's audio player loop.
115 lines
2.9 KiB
C++
115 lines
2.9 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 "ConnectionFromClient.h"
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#include "Mixer.h"
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#include <AudioServer/AudioClientEndpoint.h>
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#include <LibAudio/Queue.h>
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namespace AudioServer {
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static HashMap<int, RefPtr<ConnectionFromClient>> s_connections;
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void ConnectionFromClient::for_each(Function<void(ConnectionFromClient&)> callback)
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{
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Vector<NonnullRefPtr<ConnectionFromClient>> connections;
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for (auto& it : s_connections)
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connections.append(*it.value);
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for (auto& connection : connections)
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callback(connection);
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}
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ConnectionFromClient::ConnectionFromClient(NonnullOwnPtr<Core::LocalSocket> client_socket, int client_id, Mixer& mixer)
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: IPC::ConnectionFromClient<AudioClientEndpoint, AudioServerEndpoint>(*this, move(client_socket), client_id)
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, m_mixer(mixer)
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{
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s_connections.set(client_id, *this);
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}
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void ConnectionFromClient::die()
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{
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s_connections.remove(client_id());
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}
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void ConnectionFromClient::set_buffer(Audio::AudioQueue const& buffer)
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{
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if (!buffer.is_valid()) {
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did_misbehave("Received an invalid buffer");
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return;
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}
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if (!m_queue) {
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m_queue = m_mixer.create_queue(*this);
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if (m_saved_sample_rate.has_value())
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m_queue->set_sample_rate(m_saved_sample_rate.release_value());
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}
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// This is ugly but we know nobody uses the buffer afterwards anyways.
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m_queue->set_buffer(make<Audio::AudioQueue>(move(const_cast<Audio::AudioQueue&>(buffer))));
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}
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void ConnectionFromClient::did_change_client_volume(Badge<ClientAudioStream>, double volume)
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{
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async_client_volume_changed(volume);
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}
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Messages::AudioServer::GetSelfSampleRateResponse ConnectionFromClient::get_self_sample_rate()
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{
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if (m_queue)
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return { m_queue->sample_rate() };
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// Fall back to device sample rate since that would mean no resampling.
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return m_mixer.audiodevice_get_sample_rate();
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}
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void ConnectionFromClient::set_self_sample_rate(u32 sample_rate)
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{
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if (m_queue)
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m_queue->set_sample_rate(sample_rate);
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else
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m_saved_sample_rate = sample_rate;
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}
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Messages::AudioServer::GetSelfVolumeResponse ConnectionFromClient::get_self_volume()
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{
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return m_queue->volume().target();
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}
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void ConnectionFromClient::set_self_volume(double volume)
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{
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if (m_queue)
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m_queue->set_volume(volume);
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}
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void ConnectionFromClient::start_playback()
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{
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if (m_queue)
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m_queue->set_paused(false);
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}
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void ConnectionFromClient::pause_playback()
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{
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if (m_queue)
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m_queue->set_paused(true);
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}
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void ConnectionFromClient::clear_buffer()
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{
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if (m_queue)
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m_queue->clear();
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}
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Messages::AudioServer::IsSelfMutedResponse ConnectionFromClient::is_self_muted()
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{
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if (m_queue)
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return m_queue->is_muted();
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return false;
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}
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void ConnectionFromClient::set_self_muted(bool muted)
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{
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if (m_queue)
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m_queue->set_muted(muted);
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}
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}
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