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https://github.com/SerenityOS/serenity
synced 2024-07-22 02:26:11 +00:00
AudioServer+Userland: Decouple client sample rates from device rate
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.
This commit is contained in:
kleines Filmröllchen
Linus Groh
parent
d52a2ff10e
commit
b4fbd30b70
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@ -30,7 +30,7 @@ There are two commands available: `get` reports the state of audio variables, an
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The available variables are:
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* `(v)olume`: Audio server volume, in percent. Integer value.
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* `(m)ute`: Mute state. Boolean value, may be set with `0`, `false` or `1`, `true`.
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* `sample(r)ate`: Sample rate of the sound card. **Attention:** Most audio applications need to be restarted after changing the sample rate. Integer value.
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* `sample(r)ate`: Sample rate of the sound card. Integer value.
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Both commands and arguments can be abbreviated: Commands by their first letter, arguments by the letter in parenthesis.
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@ -163,7 +163,7 @@ private:
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case AudioTask::Type::Seek:
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VERIFY(task.data.has_value());
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m_position = Web::Platform::AudioCodecPlugin::set_loader_position(m_loader, *task.data, m_duration, audio_output->format().sampleRate());
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m_position = Web::Platform::AudioCodecPlugin::set_loader_position(m_loader, *task.data, m_duration);
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if (paused == Paused::Yes)
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Q_EMIT playback_position_updated(m_position);
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@ -211,10 +211,10 @@ private:
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auto channel_count = audio_output.format().channelCount();
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auto samples_to_load = bytes_available / bytes_per_sample / channel_count;
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auto samples = TRY(Web::Platform::AudioCodecPlugin::read_samples_from_loader(*m_loader, samples_to_load, audio_output.format().sampleRate()));
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auto samples = TRY(Web::Platform::AudioCodecPlugin::read_samples_from_loader(*m_loader, samples_to_load));
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enqueue_samples(audio_output, io_device, move(samples));
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m_position = Web::Platform::AudioCodecPlugin::current_loader_position(m_loader, audio_output.format().sampleRate());
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m_position = Web::Platform::AudioCodecPlugin::current_loader_position(m_loader);
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return Paused::No;
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}
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@ -17,7 +17,6 @@
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#include <AK/Time.h>
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#include <LibAudio/ConnectionToServer.h>
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#include <LibAudio/Queue.h>
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#include <LibAudio/Resampler.h>
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#include <LibAudio/Sample.h>
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#include <LibIPC/Connection.h>
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#include <LibThreading/Thread.h>
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@ -59,11 +58,7 @@ AudioPlayerLoop::AudioPlayerLoop(TrackManager& track_manager, Atomic<bool>& need
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, m_wav_writer(wav_writer)
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{
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m_audio_client = Audio::ConnectionToServer::try_create().release_value_but_fixme_should_propagate_errors();
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auto target_sample_rate = m_audio_client->get_sample_rate();
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if (target_sample_rate == 0)
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target_sample_rate = Music::sample_rate;
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m_resampler = Audio::ResampleHelper<DSP::Sample>(Music::sample_rate, target_sample_rate);
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m_audio_client->set_self_sample_rate(sample_rate);
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MUST(m_pipeline_thread->set_priority(sched_get_priority_max(0)));
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m_pipeline_thread->start();
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@ -108,15 +103,12 @@ intptr_t AudioPlayerLoop::pipeline_thread_main()
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ErrorOr<void> AudioPlayerLoop::send_audio_to_server()
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{
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TRY(m_resampler->try_resample_into_end(m_remaining_samples, m_buffer));
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auto sample_rate = static_cast<double>(m_resampler->target());
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auto buffer_play_time_ns = 1'000'000'000.0 / (sample_rate / static_cast<double>(Audio::AUDIO_BUFFER_SIZE));
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auto good_sleep_time = Duration::from_nanoseconds(static_cast<unsigned>(buffer_play_time_ns)).to_timespec();
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size_t start_of_chunk_to_write = 0;
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while (start_of_chunk_to_write + Audio::AUDIO_BUFFER_SIZE <= m_remaining_samples.size()) {
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auto const exact_chunk = m_remaining_samples.span().slice(start_of_chunk_to_write, Audio::AUDIO_BUFFER_SIZE);
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while (start_of_chunk_to_write + Audio::AUDIO_BUFFER_SIZE <= m_buffer.size()) {
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auto const exact_chunk = m_buffer.span().slice(start_of_chunk_to_write, Audio::AUDIO_BUFFER_SIZE);
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auto exact_chunk_array = Array<Audio::Sample, Audio::AUDIO_BUFFER_SIZE>::from_span(exact_chunk);
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TRY(m_audio_client->blocking_realtime_enqueue(exact_chunk_array, [&]() {
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@ -125,8 +117,8 @@ ErrorOr<void> AudioPlayerLoop::send_audio_to_server()
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start_of_chunk_to_write += Audio::AUDIO_BUFFER_SIZE;
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}
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m_remaining_samples.remove(0, start_of_chunk_to_write);
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VERIFY(m_remaining_samples.size() < Audio::AUDIO_BUFFER_SIZE);
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// The buffer has to have been constructed with a size of an integer multiple of the audio buffer size.
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VERIFY(start_of_chunk_to_write == m_buffer.size());
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return {};
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}
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@ -39,10 +39,8 @@ private:
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TrackManager& m_track_manager;
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FixedArray<DSP::Sample> m_buffer;
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Optional<Audio::ResampleHelper<DSP::Sample>> m_resampler;
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RefPtr<Audio::ConnectionToServer> m_audio_client;
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NonnullRefPtr<Threading::Thread> m_pipeline_thread;
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Vector<Audio::Sample, Audio::AUDIO_BUFFER_SIZE> m_remaining_samples {};
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Atomic<bool> m_should_play_audio { true };
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Atomic<bool> m_exit_requested { false };
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@ -8,6 +8,7 @@
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#pragma once
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#include <AK/Types.h>
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#include <LibAudio/Queue.h>
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#include <LibGfx/Color.h>
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namespace Music {
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@ -23,8 +24,7 @@ struct Sample {
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i16 right;
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};
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// HACK: needs to increase with device sample rate, but all of the sample_count stuff is static for now
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constexpr int sample_count = 1 << 10;
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constexpr int sample_count = Audio::AUDIO_BUFFER_SIZE * 10;
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constexpr double sample_rate = 44100;
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@ -16,7 +16,6 @@ PlaybackManager::PlaybackManager(NonnullRefPtr<Audio::ConnectionToServer> connec
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return;
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next_buffer();
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}).release_value_but_fixme_should_propagate_errors();
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m_device_sample_rate = connection->get_sample_rate();
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}
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void PlaybackManager::set_loader(NonnullRefPtr<Audio::Loader>&& loader)
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@ -24,10 +23,9 @@ void PlaybackManager::set_loader(NonnullRefPtr<Audio::Loader>&& loader)
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stop();
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m_loader = loader;
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if (m_loader) {
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m_connection->set_self_sample_rate(loader->sample_rate());
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m_total_length = m_loader->total_samples() / static_cast<float>(m_loader->sample_rate());
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m_device_samples_per_buffer = PlaybackManager::buffer_size_ms / 1000.0f * m_device_sample_rate;
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m_samples_to_load_per_buffer = PlaybackManager::buffer_size_ms / 1000.0f * m_loader->sample_rate();
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m_resampler = Audio::ResampleHelper<Audio::Sample>(m_loader->sample_rate(), m_device_sample_rate);
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m_timer->start();
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} else {
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m_timer->stop();
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@ -102,7 +100,7 @@ void PlaybackManager::next_buffer()
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if (m_paused)
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return;
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while (m_connection->remaining_samples() < m_device_samples_per_buffer * always_enqueued_buffer_count) {
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while (m_connection->remaining_samples() < m_samples_to_load_per_buffer * always_enqueued_buffer_count) {
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bool all_samples_loaded = (m_loader->loaded_samples() >= m_loader->total_samples());
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bool audio_server_done = (m_connection->remaining_samples() == 0);
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@ -121,12 +119,6 @@ void PlaybackManager::next_buffer()
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}
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auto buffer = buffer_or_error.release_value();
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m_current_buffer.swap(buffer);
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VERIFY(m_resampler.has_value());
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m_resampler->reset();
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// FIXME: Handle OOM better.
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auto resampled = MUST(FixedArray<Audio::Sample>::create(m_resampler->resample(move(m_current_buffer)).span()));
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m_current_buffer.swap(resampled);
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MUST(m_connection->async_enqueue(m_current_buffer));
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}
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}
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@ -29,7 +29,6 @@ public:
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bool toggle_pause();
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void set_loader(NonnullRefPtr<Audio::Loader>&&);
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RefPtr<Audio::Loader> loader() const { return m_loader; }
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size_t device_sample_rate() const { return m_device_sample_rate; }
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bool is_paused() const { return m_paused; }
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float total_length() const { return m_total_length; }
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@ -50,13 +49,10 @@ private:
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bool m_paused { true };
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bool m_loop = { false };
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float m_total_length { 0 };
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size_t m_device_sample_rate { 44100 };
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size_t m_device_samples_per_buffer { 0 };
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size_t m_samples_to_load_per_buffer { 0 };
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RefPtr<Audio::Loader> m_loader { nullptr };
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NonnullRefPtr<Audio::ConnectionToServer> m_connection;
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FixedArray<Audio::Sample> m_current_buffer;
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Optional<Audio::ResampleHelper<Audio::Sample>> m_resampler;
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RefPtr<Core::Timer> m_timer;
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// Controls the GUI update rate. A smaller value makes the visualizations nicer.
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@ -16,13 +16,11 @@ Player::Player(Audio::ConnectionToServer& audio_client_connection)
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m_playback_manager.on_update = [&]() {
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auto samples_played = m_playback_manager.loader()->loaded_samples();
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auto sample_rate = m_playback_manager.loader()->sample_rate();
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float source_to_dest_ratio = static_cast<float>(sample_rate) / m_playback_manager.device_sample_rate();
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samples_played *= source_to_dest_ratio;
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auto played_seconds = samples_played / sample_rate;
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time_elapsed(played_seconds);
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if (play_state() == PlayState::Playing)
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sound_buffer_played(m_playback_manager.current_buffer(), m_playback_manager.device_sample_rate(), samples_played);
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sound_buffer_played(m_playback_manager.current_buffer(), sample_rate, samples_played);
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};
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m_playback_manager.on_finished_playing = [&]() {
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set_play_state(PlayState::Stopped);
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@ -64,8 +62,7 @@ void Player::play_file_path(DeprecatedString const& path)
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m_loaded_filename = path;
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// TODO: The combination of sample count, sample rate, and sample data should be properly abstracted for the source and the playback device.
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total_samples_changed(loader->total_samples() * (static_cast<float>(loader->sample_rate()) / m_playback_manager.device_sample_rate()));
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total_samples_changed(loader->total_samples());
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m_playback_manager.set_loader(move(loader));
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file_name_changed(path);
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@ -156,11 +153,6 @@ void Player::toggle_mute()
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void Player::seek(int sample)
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{
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auto loader = m_playback_manager.loader();
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if (loader.is_null()) {
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return;
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}
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sample *= (m_playback_manager.device_sample_rate() / static_cast<float>(loader->sample_rate()));
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m_playback_manager.seek(sample);
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}
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@ -16,6 +16,7 @@
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#include <LibAudio/UserSampleQueue.h>
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#include <LibCore/Event.h>
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#include <LibThreading/Mutex.h>
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#include <Userland/Services/AudioServer/AudioClientEndpoint.h>
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#include <sched.h>
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#include <time.h>
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@ -37,6 +38,7 @@ ConnectionToServer::ConnectionToServer(NonnullOwnPtr<Core::LocalSocket> socket)
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return (intptr_t) nullptr;
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}))
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{
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update_good_sleep_time();
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async_pause_playback();
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set_buffer(*m_buffer);
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}
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@ -64,12 +66,12 @@ ErrorOr<void> ConnectionToServer::async_enqueue(FixedArray<Sample>&& samples)
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{
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if (!m_background_audio_enqueuer->is_started()) {
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m_background_audio_enqueuer->start();
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// Wait until the enqueuer has constructed its loop. A pseudo-spinlock is fine since this happens as soon as the other thread gets scheduled.
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while (!m_enqueuer_loop)
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usleep(1);
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TRY(m_background_audio_enqueuer->set_priority(THREAD_PRIORITY_MAX));
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}
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update_good_sleep_time();
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m_user_queue->append(move(samples));
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// Wake the background thread to make sure it starts enqueuing audio.
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m_enqueuer_loop->post_event(*this, make<Core::CustomEvent>(0));
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@ -86,12 +88,18 @@ void ConnectionToServer::clear_client_buffer()
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void ConnectionToServer::update_good_sleep_time()
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{
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auto sample_rate = static_cast<double>(get_sample_rate());
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auto sample_rate = static_cast<double>(get_self_sample_rate());
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auto buffer_play_time_ns = 1'000'000'000.0 / (sample_rate / static_cast<double>(AUDIO_BUFFER_SIZE));
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// A factor of 1 should be good for now.
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m_good_sleep_time = Duration::from_nanoseconds(static_cast<unsigned>(buffer_play_time_ns)).to_timespec();
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}
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void ConnectionToServer::set_self_sample_rate(u32 sample_rate)
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{
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IPC::ConnectionToServer<AudioClientEndpoint, AudioServerEndpoint>::set_self_sample_rate(sample_rate);
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update_good_sleep_time();
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}
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// Non-realtime audio writing loop
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void ConnectionToServer::custom_event(Core::CustomEvent&)
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{
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@ -55,6 +55,8 @@ public:
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// Non-realtime code needn't worry about this.
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size_t remaining_buffers() const;
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void set_self_sample_rate(u32 sample_rate);
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virtual void die() override;
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Function<void(double volume)> on_client_volume_change;
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@ -67,7 +69,6 @@ private:
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// We use this to perform the audio enqueuing on the background thread's event loop
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virtual void custom_event(Core::CustomEvent&) override;
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// FIXME: This should be called every time the sample rate changes, but we just cautiously call it on every non-realtime enqueue.
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void update_good_sleep_time();
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// Shared audio buffer: both server and client constantly read and write to/from this.
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@ -28,7 +28,7 @@ ErrorOr<NonnullOwnPtr<AudioCodecPlugin>> AudioCodecPlugin::create(NonnullRefPtr<
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return s_creation_hook(move(loader));
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}
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ErrorOr<FixedArray<Audio::Sample>> AudioCodecPlugin::read_samples_from_loader(Audio::Loader& loader, size_t samples_to_load, size_t device_sample_rate)
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ErrorOr<FixedArray<Audio::Sample>> AudioCodecPlugin::read_samples_from_loader(Audio::Loader& loader, size_t samples_to_load)
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{
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auto buffer_or_error = loader.get_more_samples(samples_to_load);
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if (buffer_or_error.is_error()) {
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@ -36,30 +36,26 @@ ErrorOr<FixedArray<Audio::Sample>> AudioCodecPlugin::read_samples_from_loader(Au
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return Error::from_string_literal("Error while loading samples");
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}
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Audio::ResampleHelper<Audio::Sample> resampler(loader.sample_rate(), device_sample_rate);
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return FixedArray<Audio::Sample>::create(resampler.resample(buffer_or_error.release_value()).span());
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return buffer_or_error.release_value();
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}
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Duration AudioCodecPlugin::set_loader_position(Audio::Loader& loader, double position, Duration duration, size_t device_sample_rate)
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Duration AudioCodecPlugin::set_loader_position(Audio::Loader& loader, double position, Duration duration)
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{
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if (loader.total_samples() == 0)
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return current_loader_position(loader, device_sample_rate);
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return current_loader_position(loader);
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auto duration_value = static_cast<double>(duration.to_milliseconds()) / 1000.0;
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position = position / duration_value * static_cast<double>(loader.total_samples() - 1);
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loader.seek(static_cast<int>(position)).release_value_but_fixme_should_propagate_errors();
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return current_loader_position(loader, device_sample_rate);
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return current_loader_position(loader);
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}
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Duration AudioCodecPlugin::current_loader_position(Audio::Loader const& loader, size_t device_sample_rate)
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Duration AudioCodecPlugin::current_loader_position(Audio::Loader const& loader)
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{
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auto samples_played = static_cast<double>(loader.loaded_samples());
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auto sample_rate = static_cast<double>(loader.sample_rate());
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auto source_to_device_ratio = sample_rate / static_cast<double>(device_sample_rate);
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samples_played *= source_to_device_ratio;
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return Duration::from_milliseconds(static_cast<i64>(samples_played / sample_rate * 1000.0));
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}
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@ -23,9 +23,9 @@ public:
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virtual ~AudioCodecPlugin();
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static ErrorOr<FixedArray<Audio::Sample>> read_samples_from_loader(Audio::Loader&, size_t samples_to_load, size_t device_sample_rate);
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static Duration set_loader_position(Audio::Loader&, double position, Duration duration, size_t device_sample_rate);
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static Duration current_loader_position(Audio::Loader const&, size_t device_sample_rate);
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static ErrorOr<FixedArray<Audio::Sample>> read_samples_from_loader(Audio::Loader&, size_t samples_to_load);
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static Duration set_loader_position(Audio::Loader&, double position, Duration duration);
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static Duration current_loader_position(Audio::Loader const&);
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virtual void resume_playback() = 0;
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virtual void pause_playback() = 0;
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@ -8,9 +8,8 @@ endpoint AudioServer
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get_self_volume() => (double volume)
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set_self_volume(double volume) => ()
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// FIXME: Decouple client sample rate from device sample rate, then remove this API
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get_sample_rate() => (u32 sample_rate)
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set_self_sample_rate(u32 sample_rate) => ()
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get_self_sample_rate() => (u32 sample_rate)
|
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// Buffer playback
|
||||
set_buffer(Audio::AudioQueue buffer) => ()
|
||||
clear_buffer() =|
|
||||
|
|
|
|||
|
|
@ -40,8 +40,11 @@ void ConnectionFromClient::set_buffer(Audio::AudioQueue const& buffer)
|
|||
did_misbehave("Received an invalid buffer");
|
||||
return;
|
||||
}
|
||||
if (!m_queue)
|
||||
if (!m_queue) {
|
||||
m_queue = m_mixer.create_queue(*this);
|
||||
if (m_saved_sample_rate.has_value())
|
||||
m_queue->set_sample_rate(m_saved_sample_rate.release_value());
|
||||
}
|
||||
|
||||
// This is ugly but we know nobody uses the buffer afterwards anyways.
|
||||
m_queue->set_buffer(make<Audio::AudioQueue>(move(const_cast<Audio::AudioQueue&>(buffer))));
|
||||
|
|
@ -52,9 +55,19 @@ void ConnectionFromClient::did_change_client_volume(Badge<ClientAudioStream>, do
|
|||
async_client_volume_changed(volume);
|
||||
}
|
||||
|
||||
Messages::AudioServer::GetSampleRateResponse ConnectionFromClient::get_sample_rate()
|
||||
Messages::AudioServer::GetSelfSampleRateResponse ConnectionFromClient::get_self_sample_rate()
|
||||
{
|
||||
return { m_mixer.audiodevice_get_sample_rate() };
|
||||
if (m_queue)
|
||||
return { m_queue->sample_rate() };
|
||||
// Fall back to device sample rate since that would mean no resampling.
|
||||
return m_mixer.audiodevice_get_sample_rate();
|
||||
}
|
||||
void ConnectionFromClient::set_self_sample_rate(u32 sample_rate)
|
||||
{
|
||||
if (m_queue)
|
||||
m_queue->set_sample_rate(sample_rate);
|
||||
else
|
||||
m_saved_sample_rate = sample_rate;
|
||||
}
|
||||
|
||||
Messages::AudioServer::GetSelfVolumeResponse ConnectionFromClient::get_self_volume()
|
||||
|
|
|
|||
|
|
@ -40,11 +40,12 @@ private:
|
|||
virtual void pause_playback() override;
|
||||
virtual Messages::AudioServer::IsSelfMutedResponse is_self_muted() override;
|
||||
virtual void set_self_muted(bool) override;
|
||||
// FIXME: Decouple client sample rate from device sample rate, then remove this endpoint
|
||||
virtual Messages::AudioServer::GetSampleRateResponse get_sample_rate() override;
|
||||
virtual Messages::AudioServer::GetSelfSampleRateResponse get_self_sample_rate() override;
|
||||
virtual void set_self_sample_rate(u32 sample_rate) override;
|
||||
|
||||
Mixer& m_mixer;
|
||||
RefPtr<ClientAudioStream> m_queue;
|
||||
Optional<u32> m_saved_sample_rate {};
|
||||
};
|
||||
|
||||
}
|
||||
|
|
|
|||
|
|
@ -39,6 +39,7 @@ Mixer::Mixer(NonnullRefPtr<Core::ConfigFile> config, NonnullOwnPtr<Core::File> d
|
|||
NonnullRefPtr<ClientAudioStream> Mixer::create_queue(ConnectionFromClient& client)
|
||||
{
|
||||
auto queue = adopt_ref(*new ClientAudioStream(client));
|
||||
queue->set_sample_rate(audiodevice_get_sample_rate());
|
||||
{
|
||||
Threading::MutexLocker const locker(m_pending_mutex);
|
||||
m_pending_mixing.append(*queue);
|
||||
|
|
@ -83,7 +84,7 @@ void Mixer::mix()
|
|||
|
||||
for (auto& mixed_sample : mixed_buffer) {
|
||||
Audio::Sample sample;
|
||||
if (!queue->get_next_sample(sample))
|
||||
if (!queue->get_next_sample(sample, audiodevice_get_sample_rate()))
|
||||
break;
|
||||
if (queue->is_muted())
|
||||
continue;
|
||||
|
|
@ -155,15 +156,22 @@ int Mixer::audiodevice_set_sample_rate(u32 sample_rate)
|
|||
int code = ioctl(m_device->fd(), SOUNDCARD_IOCTL_SET_SAMPLE_RATE, sample_rate);
|
||||
if (code != 0)
|
||||
dbgln("Error while setting sample rate to {}: ioctl error: {}", sample_rate, strerror(errno));
|
||||
// Note that the effective sample rate may be different depending on device restrictions.
|
||||
// Therefore, we delete our cache, but for efficency don't immediately read the sample rate back.
|
||||
m_cached_sample_rate = {};
|
||||
return code;
|
||||
}
|
||||
|
||||
u32 Mixer::audiodevice_get_sample_rate() const
|
||||
{
|
||||
if (m_cached_sample_rate.has_value())
|
||||
return m_cached_sample_rate.value();
|
||||
u32 sample_rate = 0;
|
||||
int code = ioctl(m_device->fd(), SOUNDCARD_IOCTL_GET_SAMPLE_RATE, &sample_rate);
|
||||
if (code != 0)
|
||||
dbgln("Error while getting sample rate: ioctl error: {}", strerror(errno));
|
||||
else
|
||||
m_cached_sample_rate = sample_rate;
|
||||
return sample_rate;
|
||||
}
|
||||
|
||||
|
|
|
|||
|
|
@ -17,6 +17,7 @@
|
|||
#include <AK/RefCounted.h>
|
||||
#include <AK/WeakPtr.h>
|
||||
#include <LibAudio/Queue.h>
|
||||
#include <LibAudio/Resampler.h>
|
||||
#include <LibCore/ConfigFile.h>
|
||||
#include <LibCore/File.h>
|
||||
#include <LibCore/Timer.h>
|
||||
|
|
@ -42,7 +43,7 @@ public:
|
|||
explicit ClientAudioStream(ConnectionFromClient&);
|
||||
~ClientAudioStream() = default;
|
||||
|
||||
bool get_next_sample(Audio::Sample& sample)
|
||||
bool 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())
|
||||
|
|
@ -60,7 +61,19 @@ public:
|
|||
|
||||
return false;
|
||||
}
|
||||
m_current_audio_chunk = result.release_value();
|
||||
// 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;
|
||||
}
|
||||
|
||||
|
|
@ -90,14 +103,21 @@ public:
|
|||
void set_volume(double const volume) { m_volume = volume; }
|
||||
bool is_muted() const { return m_muted; }
|
||||
void set_muted(bool muted) { m_muted = muted; }
|
||||
u32 sample_rate() const { return m_sample_rate; }
|
||||
void 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;
|
||||
}
|
||||
|
||||
private:
|
||||
OwnPtr<Audio::AudioQueue> m_buffer;
|
||||
Array<Audio::Sample, Audio::AUDIO_BUFFER_SIZE> m_current_audio_chunk;
|
||||
Vector<Audio::Sample> m_current_audio_chunk;
|
||||
size_t m_in_chunk_location;
|
||||
|
||||
bool m_paused { true };
|
||||
bool m_muted { false };
|
||||
u32 m_sample_rate;
|
||||
|
||||
WeakPtr<ConnectionFromClient> m_client;
|
||||
FadingProperty<double> m_volume { 1 };
|
||||
|
|
@ -137,6 +157,7 @@ private:
|
|||
Threading::ConditionVariable m_mixing_necessary { m_pending_mutex };
|
||||
|
||||
NonnullOwnPtr<Core::File> m_device;
|
||||
mutable Optional<u32> m_cached_sample_rate {};
|
||||
|
||||
NonnullRefPtr<Threading::Thread> m_sound_thread;
|
||||
|
||||
|
|
|
|||
|
|
@ -38,16 +38,16 @@ AudioCodecPluginSerenity::AudioCodecPluginSerenity(NonnullRefPtr<Audio::Connecti
|
|||
auto duration = static_cast<double>(m_loader->total_samples()) / static_cast<double>(m_loader->sample_rate());
|
||||
m_duration = Duration::from_milliseconds(static_cast<i64>(duration * 1000.0));
|
||||
|
||||
m_device_sample_rate = m_connection->get_sample_rate();
|
||||
m_device_samples_per_buffer = static_cast<size_t>(BUFFER_SIZE_MS / 1000.0 * static_cast<double>(m_device_sample_rate));
|
||||
m_samples_to_load_per_buffer = static_cast<size_t>(BUFFER_SIZE_MS / 1000.0 * static_cast<double>(m_loader->sample_rate()));
|
||||
|
||||
m_connection->set_self_sample_rate(m_loader->sample_rate());
|
||||
}
|
||||
|
||||
AudioCodecPluginSerenity::~AudioCodecPluginSerenity() = default;
|
||||
|
||||
ErrorOr<void> AudioCodecPluginSerenity::play_next_samples()
|
||||
{
|
||||
while (m_connection->remaining_samples() < m_device_samples_per_buffer * ALWAYS_ENQUEUED_BUFFER_COUNT) {
|
||||
while (m_connection->remaining_samples() < m_samples_to_load_per_buffer * ALWAYS_ENQUEUED_BUFFER_COUNT) {
|
||||
bool all_samples_loaded = m_loader->loaded_samples() >= m_loader->total_samples();
|
||||
bool audio_server_done = m_connection->remaining_samples() == 0;
|
||||
|
||||
|
|
@ -62,10 +62,10 @@ ErrorOr<void> AudioCodecPluginSerenity::play_next_samples()
|
|||
break;
|
||||
}
|
||||
|
||||
auto samples = TRY(read_samples_from_loader(m_loader, m_samples_to_load_per_buffer, m_device_sample_rate));
|
||||
auto samples = TRY(read_samples_from_loader(m_loader, m_samples_to_load_per_buffer));
|
||||
TRY(m_connection->async_enqueue(move(samples)));
|
||||
|
||||
m_position = current_loader_position(m_loader, m_device_sample_rate);
|
||||
m_position = current_loader_position(m_loader);
|
||||
}
|
||||
|
||||
return {};
|
||||
|
|
@ -90,7 +90,7 @@ void AudioCodecPluginSerenity::set_volume(double volume)
|
|||
|
||||
void AudioCodecPluginSerenity::seek(double position)
|
||||
{
|
||||
m_position = set_loader_position(m_loader, position, m_duration, m_device_sample_rate);
|
||||
m_position = set_loader_position(m_loader, position, m_duration);
|
||||
|
||||
if (on_playback_position_updated)
|
||||
on_playback_position_updated(m_position);
|
||||
|
|
|
|||
|
|
@ -40,8 +40,6 @@ private:
|
|||
Duration m_duration;
|
||||
Duration m_position;
|
||||
|
||||
size_t m_device_sample_rate { 0 };
|
||||
size_t m_device_samples_per_buffer { 0 };
|
||||
size_t m_samples_to_load_per_buffer { 0 };
|
||||
};
|
||||
|
||||
|
|
|
|||
|
|
@ -8,7 +8,6 @@
|
|||
#include <AK/Types.h>
|
||||
#include <LibAudio/ConnectionToServer.h>
|
||||
#include <LibAudio/Loader.h>
|
||||
#include <LibAudio/Resampler.h>
|
||||
#include <LibCore/ArgsParser.h>
|
||||
#include <LibCore/EventLoop.h>
|
||||
#include <LibCore/System.h>
|
||||
|
|
@ -17,8 +16,6 @@
|
|||
#include <math.h>
|
||||
#include <stdio.h>
|
||||
|
||||
// The Kernel has issues with very large anonymous buffers.
|
||||
// FIXME: This appears to be fine for now, but it's really a hack.
|
||||
constexpr size_t LOAD_CHUNK_SIZE = 128 * KiB;
|
||||
|
||||
ErrorOr<int> serenity_main(Main::Arguments arguments)
|
||||
|
|
@ -62,10 +59,7 @@ ErrorOr<int> serenity_main(Main::Arguments arguments)
|
|||
loader->num_channels() == 1 ? "Mono" : "Stereo");
|
||||
out("\033[34;1mProgress\033[0m: \033[s");
|
||||
|
||||
auto resampler = Audio::ResampleHelper<Audio::Sample>(loader->sample_rate(), audio_client->get_sample_rate());
|
||||
|
||||
// If we're downsampling, we need to appropriately load more samples at once.
|
||||
size_t const load_size = static_cast<size_t>(LOAD_CHUNK_SIZE * static_cast<double>(loader->sample_rate()) / static_cast<double>(audio_client->get_sample_rate()));
|
||||
audio_client->set_self_sample_rate(loader->sample_rate());
|
||||
|
||||
auto print_playback_update = [&]() {
|
||||
out("\033[u");
|
||||
|
|
@ -94,14 +88,12 @@ ErrorOr<int> serenity_main(Main::Arguments arguments)
|
|||
};
|
||||
|
||||
for (;;) {
|
||||
auto samples = loader->get_more_samples(load_size);
|
||||
auto samples = loader->get_more_samples(LOAD_CHUNK_SIZE);
|
||||
if (!samples.is_error()) {
|
||||
if (samples.value().size() > 0) {
|
||||
print_playback_update();
|
||||
// We can read and enqueue more samples
|
||||
resampler.reset();
|
||||
auto resampled_samples = resampler.resample(move(samples.value()));
|
||||
TRY(audio_client->async_enqueue(move(resampled_samples)));
|
||||
TRY(audio_client->async_enqueue(samples.release_value()));
|
||||
} else if (should_loop) {
|
||||
// We're done: now loop
|
||||
auto result = loader->reset();
|
||||
|
|
@ -113,7 +105,7 @@ ErrorOr<int> serenity_main(Main::Arguments arguments)
|
|||
// We're done and the server is done
|
||||
break;
|
||||
}
|
||||
while (audio_client->remaining_samples() > load_size) {
|
||||
while (audio_client->remaining_samples() > LOAD_CHUNK_SIZE) {
|
||||
// The server has enough data for now
|
||||
print_playback_update();
|
||||
usleep(1'000'000 / 10);
|
||||
|
|
|
|||
Loading…
Reference in a new issue