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https://github.com/LadybirdBrowser/ladybird.git
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AudioServer: Clean up ClientAudioStream APIs
- Use Optional references instead of pointers - Clean up some const and nullability weirdness - Use proper error return value for get_next_sample
This commit is contained in:
parent
aacb4fc590
commit
d905498fb6
Notes:
sideshowbarker
2024-07-18 03:35:30 +09:00
Author: https://github.com/kleinesfilmroellchen Commit: https://github.com/SerenityOS/serenity/commit/d905498fb6 Pull-request: https://github.com/SerenityOS/serenity/pull/20439 Reviewed-by: https://github.com/Zaggy1024 Reviewed-by: https://github.com/gmta
@ -15,9 +15,9 @@ ClientAudioStream::ClientAudioStream(ConnectionFromClient& client)
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{
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}
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ConnectionFromClient* ClientAudioStream::client()
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Optional<ConnectionFromClient&> ClientAudioStream::client()
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{
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return m_client.ptr();
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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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@ -25,14 +25,14 @@ bool ClientAudioStream::is_connected() const
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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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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 false;
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return ErrorState::ClientDisconnected;
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if (m_paused)
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return false;
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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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@ -41,30 +41,26 @@ bool ClientAudioStream::get_next_sample(Audio::Sample& sample, u32 audiodevice_s
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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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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 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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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 = attempted_resample.release_value();
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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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sample = m_current_audio_chunk[m_in_chunk_location++];
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return true;
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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(OwnPtr<Audio::AudioQueue> buffer)
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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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@ -10,7 +10,6 @@
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#include "ConnectionFromClient.h"
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#include "FadingProperty.h"
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#include <AK/Atomic.h>
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#include <AK/Badge.h>
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#include <AK/Debug.h>
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#include <AK/RefCounted.h>
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#include <AK/WeakPtr.h>
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@ -20,22 +19,29 @@ namespace AudioServer {
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class ClientAudioStream : public RefCounted<ClientAudioStream> {
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public:
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enum class ErrorState {
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ClientDisconnected,
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ClientPaused,
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ClientUnderrun,
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ResamplingError,
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};
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explicit ClientAudioStream(ConnectionFromClient&);
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~ClientAudioStream() = default;
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bool get_next_sample(Audio::Sample& sample, u32 audiodevice_sample_rate);
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ErrorOr<Audio::Sample, ErrorState> get_next_sample(u32 audiodevice_sample_rate);
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void clear();
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bool is_connected() const;
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ConnectionFromClient* client();
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Optional<ConnectionFromClient&> client();
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void set_buffer(OwnPtr<Audio::AudioQueue> buffer);
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void set_buffer(NonnullOwnPtr<Audio::AudioQueue> buffer);
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void set_paused(bool paused);
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FadingProperty<double>& volume();
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double volume() const;
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void set_volume(double const volume);
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void set_volume(double volume);
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bool is_muted() const;
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void set_muted(bool muted);
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u32 sample_rate() const;
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@ -73,18 +73,20 @@ void Mixer::mix()
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// Mix the buffers together into the output
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for (auto& queue : active_mix_queues) {
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if (!queue->client()) {
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if (!queue->client().has_value()) {
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queue->clear();
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continue;
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}
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queue->volume().advance_time();
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// FIXME: Perform sample extraction and mixing in two separate loops so they can be more easily vectorized.
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for (auto& mixed_sample : mixed_buffer) {
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Audio::Sample sample;
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if (!queue->get_next_sample(sample, audiodevice_get_sample_rate()))
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auto sample_or_error = queue->get_next_sample(audiodevice_get_sample_rate());
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if (sample_or_error.is_error())
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break;
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if (queue->is_muted())
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continue;
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auto sample = sample_or_error.release_value();
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sample.log_multiply(SAMPLE_HEADROOM);
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sample.log_multiply(static_cast<float>(queue->volume()));
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mixed_sample += sample;
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