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author | Kelebek1 <eeeedddccc@hotmail.co.uk> | 2022-07-17 00:48:45 +0200 |
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committer | Kelebek1 <eeeedddccc@hotmail.co.uk> | 2022-07-22 02:11:32 +0200 |
commit | 458da8a94877677f086f06cdeecf959ec4283a33 (patch) | |
tree | 583166d77602ad90a0d552f37de8729ad80fd6c1 /src/audio_core/sink | |
parent | Merge pull request #8598 from Link4565/recv-dontwait (diff) | |
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Diffstat (limited to 'src/audio_core/sink')
-rw-r--r-- | src/audio_core/sink/cubeb_sink.cpp | 651 | ||||
-rw-r--r-- | src/audio_core/sink/cubeb_sink.h | 110 | ||||
-rw-r--r-- | src/audio_core/sink/null_sink.h | 52 | ||||
-rw-r--r-- | src/audio_core/sink/sdl2_sink.cpp | 556 | ||||
-rw-r--r-- | src/audio_core/sink/sdl2_sink.h | 101 | ||||
-rw-r--r-- | src/audio_core/sink/sink.h | 106 | ||||
-rw-r--r-- | src/audio_core/sink/sink_details.cpp | 91 | ||||
-rw-r--r-- | src/audio_core/sink/sink_details.h | 43 | ||||
-rw-r--r-- | src/audio_core/sink/sink_stream.h | 224 |
9 files changed, 1934 insertions, 0 deletions
diff --git a/src/audio_core/sink/cubeb_sink.cpp b/src/audio_core/sink/cubeb_sink.cpp new file mode 100644 index 000000000..a4e28de6d --- /dev/null +++ b/src/audio_core/sink/cubeb_sink.cpp @@ -0,0 +1,651 @@ +// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#include <algorithm> +#include <atomic> +#include <span> + +#include "audio_core/audio_core.h" +#include "audio_core/audio_event.h" +#include "audio_core/audio_manager.h" +#include "audio_core/sink/cubeb_sink.h" +#include "audio_core/sink/sink_stream.h" +#include "common/assert.h" +#include "common/fixed_point.h" +#include "common/logging/log.h" +#include "common/reader_writer_queue.h" +#include "common/ring_buffer.h" +#include "common/settings.h" +#include "core/core.h" + +#ifdef _WIN32 +#include <objbase.h> +#undef CreateEvent +#endif + +namespace AudioCore::Sink { +/** + * Cubeb sink stream, responsible for sinking samples to hardware. + */ +class CubebSinkStream final : public SinkStream { +public: + /** + * Create a new sink stream. + * + * @param ctx_ - Cubeb context to create this stream with. + * @param device_channels_ - Number of channels supported by the hardware. + * @param system_channels_ - Number of channels the audio systems expect. + * @param output_device - Cubeb output device id. + * @param input_device - Cubeb input device id. + * @param name_ - Name of this stream. + * @param type_ - Type of this stream. + * @param system_ - Core system. + * @param event - Event used only for audio renderer, signalled on buffer consume. + */ + CubebSinkStream(cubeb* ctx_, const u32 device_channels_, const u32 system_channels_, + cubeb_devid output_device, cubeb_devid input_device, const std::string& name_, + const StreamType type_, Core::System& system_) + : ctx{ctx_}, type{type_}, system{system_} { +#ifdef _WIN32 + CoInitializeEx(nullptr, COINIT_MULTITHREADED); +#endif + name = name_; + device_channels = device_channels_; + system_channels = system_channels_; + + cubeb_stream_params params{}; + params.rate = TargetSampleRate; + params.channels = device_channels; + params.format = CUBEB_SAMPLE_S16LE; + params.prefs = CUBEB_STREAM_PREF_NONE; + switch (params.channels) { + case 1: + params.layout = CUBEB_LAYOUT_MONO; + break; + case 2: + params.layout = CUBEB_LAYOUT_STEREO; + break; + case 6: + params.layout = CUBEB_LAYOUT_3F2_LFE; + break; + } + + u32 minimum_latency{0}; + const auto latency_error = cubeb_get_min_latency(ctx, ¶ms, &minimum_latency); + if (latency_error != CUBEB_OK) { + LOG_CRITICAL(Audio_Sink, "Error getting minimum latency, error: {}", latency_error); + minimum_latency = 256U; + } + + minimum_latency = std::max(minimum_latency, 256u); + + playing_buffer.consumed = true; + + LOG_DEBUG(Service_Audio, + "Opening cubeb stream {} type {} with: rate {} channels {} (system channels {}) " + "latency {}", + name, type, params.rate, params.channels, system_channels, minimum_latency); + + auto init_error{0}; + if (type == StreamType::In) { + init_error = cubeb_stream_init(ctx, &stream_backend, name.c_str(), input_device, + ¶ms, output_device, nullptr, minimum_latency, + &CubebSinkStream::DataCallback, + &CubebSinkStream::StateCallback, this); + } else { + init_error = cubeb_stream_init(ctx, &stream_backend, name.c_str(), input_device, + nullptr, output_device, ¶ms, minimum_latency, + &CubebSinkStream::DataCallback, + &CubebSinkStream::StateCallback, this); + } + + if (init_error != CUBEB_OK) { + LOG_CRITICAL(Audio_Sink, "Error initializing cubeb stream, error: {}", init_error); + return; + } + } + + /** + * Destroy the sink stream. + */ + ~CubebSinkStream() override { + LOG_DEBUG(Service_Audio, "Destructing cubeb stream {}", name); + + if (!ctx) { + return; + } + + Finalize(); + +#ifdef _WIN32 + CoUninitialize(); +#endif + } + + /** + * Finalize the sink stream. + */ + void Finalize() override { + Stop(); + cubeb_stream_destroy(stream_backend); + } + + /** + * Start the sink stream. + * + * @param resume - Set to true if this is resuming the stream a previously-active stream. + * Default false. + */ + void Start(const bool resume = false) override { + if (!ctx) { + return; + } + + if (resume && was_playing) { + if (cubeb_stream_start(stream_backend) != CUBEB_OK) { + LOG_CRITICAL(Audio_Sink, "Error starting cubeb stream"); + } + paused = false; + } else if (!resume) { + if (cubeb_stream_start(stream_backend) != CUBEB_OK) { + LOG_CRITICAL(Audio_Sink, "Error starting cubeb stream"); + } + paused = false; + } + } + + /** + * Stop the sink stream. + */ + void Stop() override { + if (!ctx) { + return; + } + + if (cubeb_stream_stop(stream_backend) != CUBEB_OK) { + LOG_CRITICAL(Audio_Sink, "Error stopping cubeb stream"); + } + + was_playing.store(!paused); + paused = true; + } + + /** + * Append a new buffer and its samples to a waiting queue to play. + * + * @param buffer - Audio buffer information to be queued. + * @param samples - The s16 samples to be queue for playback. + */ + void AppendBuffer(::AudioCore::Sink::SinkBuffer& buffer, std::vector<s16>& samples) override { + if (type == StreamType::In) { + queue.enqueue(buffer); + queued_buffers++; + } else { + constexpr s32 min{std::numeric_limits<s16>::min()}; + constexpr s32 max{std::numeric_limits<s16>::max()}; + + auto yuzu_volume{Settings::Volume()}; + auto volume{system_volume * device_volume * yuzu_volume}; + + if (system_channels == 6 && device_channels == 2) { + // We're given 6 channels, but our device only outputs 2, so downmix. + constexpr std::array<f32, 4> down_mix_coeff{1.0f, 0.707f, 0.251f, 0.707f}; + + for (u32 read_index = 0, write_index = 0; read_index < samples.size(); + read_index += system_channels, write_index += device_channels) { + const auto left_sample{ + ((Common::FixedPoint<49, 15>( + samples[read_index + static_cast<u32>(Channels::FrontLeft)]) * + down_mix_coeff[0] + + samples[read_index + static_cast<u32>(Channels::Center)] * + down_mix_coeff[1] + + samples[read_index + static_cast<u32>(Channels::LFE)] * + down_mix_coeff[2] + + samples[read_index + static_cast<u32>(Channels::BackLeft)] * + down_mix_coeff[3]) * + volume) + .to_int()}; + + const auto right_sample{ + ((Common::FixedPoint<49, 15>( + samples[read_index + static_cast<u32>(Channels::FrontRight)]) * + down_mix_coeff[0] + + samples[read_index + static_cast<u32>(Channels::Center)] * + down_mix_coeff[1] + + samples[read_index + static_cast<u32>(Channels::LFE)] * + down_mix_coeff[2] + + samples[read_index + static_cast<u32>(Channels::BackRight)] * + down_mix_coeff[3]) * + volume) + .to_int()}; + + samples[write_index + static_cast<u32>(Channels::FrontLeft)] = + static_cast<s16>(std::clamp(left_sample, min, max)); + samples[write_index + static_cast<u32>(Channels::FrontRight)] = + static_cast<s16>(std::clamp(right_sample, min, max)); + } + + samples.resize(samples.size() / system_channels * device_channels); + + } else if (system_channels == 2 && device_channels == 6) { + // We need moar samples! Not all games will provide 6 channel audio. + // TODO: Implement some upmixing here. Currently just passthrough, with other + // channels left as silence. + std::vector<s16> new_samples(samples.size() / system_channels * device_channels, 0); + + for (u32 read_index = 0, write_index = 0; read_index < samples.size(); + read_index += system_channels, write_index += device_channels) { + const auto left_sample{static_cast<s16>(std::clamp( + static_cast<s32>( + static_cast<f32>( + samples[read_index + static_cast<u32>(Channels::FrontLeft)]) * + volume), + min, max))}; + + new_samples[write_index + static_cast<u32>(Channels::FrontLeft)] = left_sample; + + const auto right_sample{static_cast<s16>(std::clamp( + static_cast<s32>( + static_cast<f32>( + samples[read_index + static_cast<u32>(Channels::FrontRight)]) * + volume), + min, max))}; + + new_samples[write_index + static_cast<u32>(Channels::FrontRight)] = + right_sample; + } + samples = std::move(new_samples); + + } else if (volume != 1.0f) { + for (u32 i = 0; i < samples.size(); i++) { + samples[i] = static_cast<s16>(std::clamp( + static_cast<s32>(static_cast<f32>(samples[i]) * volume), min, max)); + } + } + + samples_buffer.Push(samples); + queue.enqueue(buffer); + queued_buffers++; + } + } + + /** + * Release a buffer. Audio In only, will fill a buffer with recorded samples. + * + * @param num_samples - Maximum number of samples to receive. + * @return Vector of recorded samples. May have fewer than num_samples. + */ + std::vector<s16> ReleaseBuffer(const u64 num_samples) override { + static constexpr s32 min = std::numeric_limits<s16>::min(); + static constexpr s32 max = std::numeric_limits<s16>::max(); + + auto samples{samples_buffer.Pop(num_samples)}; + + // TODO: Up-mix to 6 channels if the game expects it. + // For audio input this is unlikely to ever be the case though. + + // Incoming mic volume seems to always be very quiet, so multiply by an additional 8 here. + // TODO: Play with this and find something that works better. + auto volume{system_volume * device_volume * 8}; + for (u32 i = 0; i < samples.size(); i++) { + samples[i] = static_cast<s16>( + std::clamp(static_cast<s32>(static_cast<f32>(samples[i]) * volume), min, max)); + } + + if (samples.size() < num_samples) { + samples.resize(num_samples, 0); + } + return samples; + } + + /** + * Check if a certain buffer has been consumed (fully played). + * + * @param tag - Unique tag of a buffer to check for. + * @return True if the buffer has been played, otherwise false. + */ + bool IsBufferConsumed(const u64 tag) override { + if (released_buffer.tag == 0) { + if (!released_buffers.try_dequeue(released_buffer)) { + return false; + } + } + + if (released_buffer.tag == tag) { + released_buffer.tag = 0; + return true; + } + return false; + } + + /** + * Empty out the buffer queue. + */ + void ClearQueue() override { + samples_buffer.Pop(); + while (queue.pop()) { + } + while (released_buffers.pop()) { + } + queued_buffers = 0; + released_buffer = {}; + playing_buffer = {}; + playing_buffer.consumed = true; + } + +private: + /** + * Signal events back to the audio system that a buffer was played/can be filled. + * + * @param buffer - Consumed audio buffer to be released. + */ + void SignalEvent(const ::AudioCore::Sink::SinkBuffer& buffer) { + auto& manager{system.AudioCore().GetAudioManager()}; + switch (type) { + case StreamType::Out: + released_buffers.enqueue(buffer); + manager.SetEvent(Event::Type::AudioOutManager, true); + break; + case StreamType::In: + released_buffers.enqueue(buffer); + manager.SetEvent(Event::Type::AudioInManager, true); + break; + case StreamType::Render: + break; + } + } + + /** + * Main callback from Cubeb. Either expects samples from us (audio render/audio out), or will + * provide samples to be copied (audio in). + * + * @param stream - Cubeb-specific data about the stream. + * @param user_data - Custom data pointer passed along, points to a CubebSinkStream. + * @param in_buff - Input buffer to be used if the stream is an input type. + * @param out_buff - Output buffer to be used if the stream is an output type. + * @param num_frames_ - Number of frames of audio in the buffers. Note: Not number of samples. + */ + static long DataCallback([[maybe_unused]] cubeb_stream* stream, void* user_data, + [[maybe_unused]] const void* in_buff, void* out_buff, + long num_frames_) { + auto* impl = static_cast<CubebSinkStream*>(user_data); + if (!impl) { + return -1; + } + + const std::size_t num_channels = impl->GetDeviceChannels(); + const std::size_t frame_size = num_channels; + const std::size_t frame_size_bytes = frame_size * sizeof(s16); + const std::size_t num_frames{static_cast<size_t>(num_frames_)}; + size_t frames_written{0}; + [[maybe_unused]] bool underrun{false}; + + if (impl->type == StreamType::In) { + // INPUT + std::span<const s16> input_buffer{reinterpret_cast<const s16*>(in_buff), + num_frames * frame_size}; + + while (frames_written < num_frames) { + auto& playing_buffer{impl->playing_buffer}; + + // If the playing buffer has been consumed or has no frames, we need a new one + if (playing_buffer.consumed || playing_buffer.frames == 0) { + if (!impl->queue.try_dequeue(impl->playing_buffer)) { + // If no buffer was available we've underrun, just push the samples and + // continue. + underrun = true; + impl->samples_buffer.Push(&input_buffer[frames_written * frame_size], + (num_frames - frames_written) * frame_size); + frames_written = num_frames; + continue; + } else { + // Successfully got a new buffer, mark the old one as consumed and signal. + impl->queued_buffers--; + impl->SignalEvent(impl->playing_buffer); + } + } + + // Get the minimum frames available between the currently playing buffer, and the + // amount we have left to fill + size_t frames_available{ + std::min(playing_buffer.frames - playing_buffer.frames_played, + num_frames - frames_written)}; + + impl->samples_buffer.Push(&input_buffer[frames_written * frame_size], + frames_available * frame_size); + + frames_written += frames_available; + playing_buffer.frames_played += frames_available; + + // If that's all the frames in the current buffer, add its samples and mark it as + // consumed + if (playing_buffer.frames_played >= playing_buffer.frames) { + impl->AddPlayedSampleCount(playing_buffer.frames_played * num_channels); + impl->playing_buffer.consumed = true; + } + } + + std::memcpy(&impl->last_frame[0], &input_buffer[(frames_written - 1) * frame_size], + frame_size_bytes); + } else { + // OUTPUT + std::span<s16> output_buffer{reinterpret_cast<s16*>(out_buff), num_frames * frame_size}; + + while (frames_written < num_frames) { + auto& playing_buffer{impl->playing_buffer}; + + // If the playing buffer has been consumed or has no frames, we need a new one + if (playing_buffer.consumed || playing_buffer.frames == 0) { + if (!impl->queue.try_dequeue(impl->playing_buffer)) { + // If no buffer was available we've underrun, fill the remaining buffer with + // the last written frame and continue. + underrun = true; + for (size_t i = frames_written; i < num_frames; i++) { + std::memcpy(&output_buffer[i * frame_size], &impl->last_frame[0], + frame_size_bytes); + } + frames_written = num_frames; + continue; + } else { + // Successfully got a new buffer, mark the old one as consumed and signal. + impl->queued_buffers--; + impl->SignalEvent(impl->playing_buffer); + } + } + + // Get the minimum frames available between the currently playing buffer, and the + // amount we have left to fill + size_t frames_available{ + std::min(playing_buffer.frames - playing_buffer.frames_played, + num_frames - frames_written)}; + + impl->samples_buffer.Pop(&output_buffer[frames_written * frame_size], + frames_available * frame_size); + + frames_written += frames_available; + playing_buffer.frames_played += frames_available; + + // If that's all the frames in the current buffer, add its samples and mark it as + // consumed + if (playing_buffer.frames_played >= playing_buffer.frames) { + impl->AddPlayedSampleCount(playing_buffer.frames_played * num_channels); + impl->playing_buffer.consumed = true; + } + } + + std::memcpy(&impl->last_frame[0], &output_buffer[(frames_written - 1) * frame_size], + frame_size_bytes); + } + + return num_frames_; + } + + /** + * Cubeb callback for if a device state changes. Unused currently. + * + * @param stream - Cubeb-specific data about the stream. + * @param user_data - Custom data pointer passed along, points to a CubebSinkStream. + * @param state - New state of the device. + */ + static void StateCallback([[maybe_unused]] cubeb_stream* stream, + [[maybe_unused]] void* user_data, + [[maybe_unused]] cubeb_state state) {} + + /// Main Cubeb context + cubeb* ctx{}; + /// Cubeb stream backend + cubeb_stream* stream_backend{}; + /// Name of this stream + std::string name{}; + /// Type of this stream + StreamType type; + /// Core system + Core::System& system; + /// Ring buffer of the samples waiting to be played or consumed + Common::RingBuffer<s16, 0x10000> samples_buffer; + /// Audio buffers queued and waiting to play + Common::ReaderWriterQueue<::AudioCore::Sink::SinkBuffer> queue; + /// The currently-playing audio buffer + ::AudioCore::Sink::SinkBuffer playing_buffer{}; + /// Audio buffers which have been played and are in queue to be released by the audio system + Common::ReaderWriterQueue<::AudioCore::Sink::SinkBuffer> released_buffers{}; + /// Currently released buffer waiting to be taken by the audio system + ::AudioCore::Sink::SinkBuffer released_buffer{}; + /// The last played (or received) frame of audio, used when the callback underruns + std::array<s16, MaxChannels> last_frame{}; +}; + +CubebSink::CubebSink(std::string_view target_device_name) { + // Cubeb requires COM to be initialized on the thread calling cubeb_init on Windows +#ifdef _WIN32 + com_init_result = CoInitializeEx(nullptr, COINIT_MULTITHREADED); +#endif + + if (cubeb_init(&ctx, "yuzu", nullptr) != CUBEB_OK) { + LOG_CRITICAL(Audio_Sink, "cubeb_init failed"); + return; + } + + if (target_device_name != auto_device_name && !target_device_name.empty()) { + cubeb_device_collection collection; + if (cubeb_enumerate_devices(ctx, CUBEB_DEVICE_TYPE_OUTPUT, &collection) != CUBEB_OK) { + LOG_WARNING(Audio_Sink, "Audio output device enumeration not supported"); + } else { + const auto collection_end{collection.device + collection.count}; + const auto device{ + std::find_if(collection.device, collection_end, [&](const cubeb_device_info& info) { + return info.friendly_name != nullptr && + target_device_name == std::string(info.friendly_name); + })}; + if (device != collection_end) { + output_device = device->devid; + } + cubeb_device_collection_destroy(ctx, &collection); + } + } + + cubeb_get_max_channel_count(ctx, &device_channels); + device_channels = device_channels >= 6U ? 6U : 2U; +} + +CubebSink::~CubebSink() { + if (!ctx) { + return; + } + + for (auto& sink_stream : sink_streams) { + sink_stream.reset(); + } + + cubeb_destroy(ctx); + +#ifdef _WIN32 + if (SUCCEEDED(com_init_result)) { + CoUninitialize(); + } +#endif +} + +SinkStream* CubebSink::AcquireSinkStream(Core::System& system, const u32 system_channels, + const std::string& name, const StreamType type) { + SinkStreamPtr& stream = sink_streams.emplace_back(std::make_unique<CubebSinkStream>( + ctx, device_channels, system_channels, output_device, input_device, name, type, system)); + + return stream.get(); +} + +void CubebSink::CloseStream(const SinkStream* stream) { + for (size_t i = 0; i < sink_streams.size(); i++) { + if (sink_streams[i].get() == stream) { + sink_streams[i].reset(); + sink_streams.erase(sink_streams.begin() + i); + break; + } + } +} + +void CubebSink::CloseStreams() { + sink_streams.clear(); +} + +void CubebSink::PauseStreams() { + for (auto& stream : sink_streams) { + stream->Stop(); + } +} + +void CubebSink::UnpauseStreams() { + for (auto& stream : sink_streams) { + stream->Start(true); + } +} + +f32 CubebSink::GetDeviceVolume() const { + if (sink_streams.empty()) { + return 1.0f; + } + + return sink_streams[0]->GetDeviceVolume(); +} + +void CubebSink::SetDeviceVolume(const f32 volume) { + for (auto& stream : sink_streams) { + stream->SetDeviceVolume(volume); + } +} + +void CubebSink::SetSystemVolume(const f32 volume) { + for (auto& stream : sink_streams) { + stream->SetSystemVolume(volume); + } +} + +std::vector<std::string> ListCubebSinkDevices(const bool capture) { + std::vector<std::string> device_list; + cubeb* ctx; + + if (cubeb_init(&ctx, "yuzu Device Enumerator", nullptr) != CUBEB_OK) { + LOG_CRITICAL(Audio_Sink, "cubeb_init failed"); + return {}; + } + + auto type{capture ? CUBEB_DEVICE_TYPE_INPUT : CUBEB_DEVICE_TYPE_OUTPUT}; + cubeb_device_collection collection; + if (cubeb_enumerate_devices(ctx, type, &collection) != CUBEB_OK) { + LOG_WARNING(Audio_Sink, "Audio output device enumeration not supported"); + } else { + for (std::size_t i = 0; i < collection.count; i++) { + const cubeb_device_info& device = collection.device[i]; + if (device.friendly_name && device.friendly_name[0] != '\0' && + device.state == CUBEB_DEVICE_STATE_ENABLED) { + device_list.emplace_back(device.friendly_name); + } + } + cubeb_device_collection_destroy(ctx, &collection); + } + + cubeb_destroy(ctx); + return device_list; +} + +} // namespace AudioCore::Sink diff --git a/src/audio_core/sink/cubeb_sink.h b/src/audio_core/sink/cubeb_sink.h new file mode 100644 index 000000000..f0f43dfa1 --- /dev/null +++ b/src/audio_core/sink/cubeb_sink.h @@ -0,0 +1,110 @@ +// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#pragma once + +#include <string> +#include <vector> + +#include <cubeb/cubeb.h> + +#include "audio_core/sink/sink.h" + +namespace Core { +class System; +} + +namespace AudioCore::Sink { +class SinkStream; + +/** + * Cubeb backend sink, holds multiple output streams and is responsible for sinking samples to + * hardware. Used by Audio Render, Audio In and Audio Out. + */ +class CubebSink final : public Sink { +public: + explicit CubebSink(std::string_view device_id); + ~CubebSink() override; + + /** + * Create a new sink stream. + * + * @param system - Core system. + * @param system_channels - Number of channels the audio system expects. + * May differ from the device's channel count. + * @param name - Name of this stream. + * @param type - Type of this stream, render/in/out. + * @param event - Audio render only, a signal used to prevent the renderer running too + * fast. + * @return A pointer to the created SinkStream + */ + SinkStream* AcquireSinkStream(Core::System& system, u32 system_channels, + const std::string& name, StreamType type) override; + + /** + * Close a given stream. + * + * @param stream - The stream to close. + */ + void CloseStream(const SinkStream* stream) override; + + /** + * Close all streams. + */ + void CloseStreams() override; + + /** + * Pause all streams. + */ + void PauseStreams() override; + + /** + * Unpause all streams. + */ + void UnpauseStreams() override; + + /** + * Get the device volume. Set from calls to the IAudioDevice service. + * + * @return Volume of the device. + */ + f32 GetDeviceVolume() const override; + + /** + * Set the device volume. Set from calls to the IAudioDevice service. + * + * @param volume - New volume of the device. + */ + void SetDeviceVolume(f32 volume) override; + + /** + * Set the system volume. Comes from the audio system using this stream. + * + * @param volume - New volume of the system. + */ + void SetSystemVolume(f32 volume) override; + +private: + /// Backend Cubeb context + cubeb* ctx{}; + /// Cubeb id of the actual hardware output device + cubeb_devid output_device{}; + /// Cubeb id of the actual hardware input device + cubeb_devid input_device{}; + /// Vector of streams managed by this sink + std::vector<SinkStreamPtr> sink_streams{}; + +#ifdef _WIN32 + /// Cubeb required COM to be initialized multi-threaded on Windows + u32 com_init_result = 0; +#endif +}; + +/** + * Get a list of conencted devices from Cubeb. + * + * @param capture - Return input (capture) devices if true, otherwise output devices. + */ +std::vector<std::string> ListCubebSinkDevices(bool capture); + +} // namespace AudioCore::Sink diff --git a/src/audio_core/sink/null_sink.h b/src/audio_core/sink/null_sink.h new file mode 100644 index 000000000..47a342171 --- /dev/null +++ b/src/audio_core/sink/null_sink.h @@ -0,0 +1,52 @@ +// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#pragma once + +#include "audio_core/sink/sink.h" +#include "audio_core/sink/sink_stream.h" + +namespace AudioCore::Sink { +/** + * A no-op sink for when no audio out is wanted. + */ +class NullSink final : public Sink { +public: + explicit NullSink(std::string_view) {} + ~NullSink() override = default; + + SinkStream* AcquireSinkStream([[maybe_unused]] Core::System& system, + [[maybe_unused]] u32 system_channels, + [[maybe_unused]] const std::string& name, + [[maybe_unused]] StreamType type) override { + return &null_sink_stream; + } + + void CloseStream([[maybe_unused]] const SinkStream* stream) override {} + void CloseStreams() override {} + void PauseStreams() override {} + void UnpauseStreams() override {} + f32 GetDeviceVolume() const override { + return 1.0f; + } + void SetDeviceVolume(f32 volume) override {} + void SetSystemVolume(f32 volume) override {} + +private: + struct NullSinkStreamImpl final : SinkStream { + void Finalize() override {} + void Start(bool resume = false) override {} + void Stop() override {} + void AppendBuffer([[maybe_unused]] ::AudioCore::Sink::SinkBuffer& buffer, + [[maybe_unused]] std::vector<s16>& samples) override {} + std::vector<s16> ReleaseBuffer([[maybe_unused]] u64 num_samples) override { + return {}; + } + bool IsBufferConsumed([[maybe_unused]] const u64 tag) { + return true; + } + void ClearQueue() override {} + } null_sink_stream; +}; + +} // namespace AudioCore::Sink diff --git a/src/audio_core/sink/sdl2_sink.cpp b/src/audio_core/sink/sdl2_sink.cpp new file mode 100644 index 000000000..d6c9ec90d --- /dev/null +++ b/src/audio_core/sink/sdl2_sink.cpp @@ -0,0 +1,556 @@ +// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#include <algorithm> +#include <atomic> + +#include "audio_core/audio_core.h" +#include "audio_core/audio_event.h" +#include "audio_core/audio_manager.h" +#include "audio_core/sink/sdl2_sink.h" +#include "audio_core/sink/sink_stream.h" +#include "common/assert.h" +#include "common/fixed_point.h" +#include "common/logging/log.h" +#include "common/reader_writer_queue.h" +#include "common/ring_buffer.h" +#include "common/settings.h" +#include "core/core.h" + +// Ignore -Wimplicit-fallthrough due to https://github.com/libsdl-org/SDL/issues/4307 +#ifdef __clang__ +#pragma clang diagnostic push +#pragma clang diagnostic ignored "-Wimplicit-fallthrough" +#endif +#include <SDL.h> +#ifdef __clang__ +#pragma clang diagnostic pop +#endif + +namespace AudioCore::Sink { +/** + * SDL sink stream, responsible for sinking samples to hardware. + */ +class SDLSinkStream final : public SinkStream { +public: + /** + * Create a new sink stream. + * + * @param device_channels_ - Number of channels supported by the hardware. + * @param system_channels_ - Number of channels the audio systems expect. + * @param output_device - Name of the output device to use for this stream. + * @param input_device - Name of the input device to use for this stream. + * @param type_ - Type of this stream. + * @param system_ - Core system. + * @param event - Event used only for audio renderer, signalled on buffer consume. + */ + SDLSinkStream(u32 device_channels_, const u32 system_channels_, + const std::string& output_device, const std::string& input_device, + const StreamType type_, Core::System& system_) + : type{type_}, system{system_} { + system_channels = system_channels_; + device_channels = device_channels_; + + SDL_AudioSpec spec; + spec.freq = TargetSampleRate; + spec.channels = static_cast<u8>(device_channels); + spec.format = AUDIO_S16SYS; + if (type == StreamType::Render) { + spec.samples = TargetSampleCount; + } else { + spec.samples = 1024; + } + spec.callback = &SDLSinkStream::DataCallback; + spec.userdata = this; + + playing_buffer.consumed = true; + + std::string device_name{output_device}; + bool capture{false}; + if (type == StreamType::In) { + device_name = input_device; + capture = true; + } + + SDL_AudioSpec obtained; + if (device_name.empty()) { + device = SDL_OpenAudioDevice(nullptr, capture, &spec, &obtained, false); + } else { + device = SDL_OpenAudioDevice(device_name.c_str(), capture, &spec, &obtained, false); + } + + if (device == 0) { + LOG_CRITICAL(Audio_Sink, "Error opening SDL audio device: {}", SDL_GetError()); + return; + } + + LOG_DEBUG(Service_Audio, + "Opening sdl stream {} with: rate {} channels {} (system channels {}) " + " samples {}", + device, obtained.freq, obtained.channels, system_channels, obtained.samples); + } + + /** + * Destroy the sink stream. + */ + ~SDLSinkStream() override { + if (device == 0) { + return; + } + + SDL_CloseAudioDevice(device); + } + + /** + * Finalize the sink stream. + */ + void Finalize() override { + if (device == 0) { + return; + } + + SDL_CloseAudioDevice(device); + } + + /** + * Start the sink stream. + * + * @param resume - Set to true if this is resuming the stream a previously-active stream. + * Default false. + */ + void Start(const bool resume = false) override { + if (device == 0) { + return; + } + + if (resume && was_playing) { + SDL_PauseAudioDevice(device, 0); + paused = false; + } else if (!resume) { + SDL_PauseAudioDevice(device, 0); + paused = false; + } + } + + /** + * Stop the sink stream. + */ + void Stop() { + if (device == 0) { + return; + } + SDL_PauseAudioDevice(device, 1); + paused = true; + } + + /** + * Append a new buffer and its samples to a waiting queue to play. + * + * @param buffer - Audio buffer information to be queued. + * @param samples - The s16 samples to be queue for playback. + */ + void AppendBuffer(::AudioCore::Sink::SinkBuffer& buffer, std::vector<s16>& samples) override { + if (type == StreamType::In) { + queue.enqueue(buffer); + queued_buffers++; + } else { + constexpr s32 min = std::numeric_limits<s16>::min(); + constexpr s32 max = std::numeric_limits<s16>::max(); + + auto yuzu_volume{Settings::Volume()}; + auto volume{system_volume * device_volume * yuzu_volume}; + + if (system_channels == 6 && device_channels == 2) { + // We're given 6 channels, but our device only outputs 2, so downmix. + constexpr std::array<f32, 4> down_mix_coeff{1.0f, 0.707f, 0.251f, 0.707f}; + + for (u32 read_index = 0, write_index = 0; read_index < samples.size(); + read_index += system_channels, write_index += device_channels) { + const auto left_sample{ + ((Common::FixedPoint<49, 15>( + samples[read_index + static_cast<u32>(Channels::FrontLeft)]) * + down_mix_coeff[0] + + samples[read_index + static_cast<u32>(Channels::Center)] * + down_mix_coeff[1] + + samples[read_index + static_cast<u32>(Channels::LFE)] * + down_mix_coeff[2] + + samples[read_index + static_cast<u32>(Channels::BackLeft)] * + down_mix_coeff[3]) * + volume) + .to_int()}; + + const auto right_sample{ + ((Common::FixedPoint<49, 15>( + samples[read_index + static_cast<u32>(Channels::FrontRight)]) * + down_mix_coeff[0] + + samples[read_index + static_cast<u32>(Channels::Center)] * + down_mix_coeff[1] + + samples[read_index + static_cast<u32>(Channels::LFE)] * + down_mix_coeff[2] + + samples[read_index + static_cast<u32>(Channels::BackRight)] * + down_mix_coeff[3]) * + volume) + .to_int()}; + + samples[write_index + static_cast<u32>(Channels::FrontLeft)] = + static_cast<s16>(std::clamp(left_sample, min, max)); + samples[write_index + static_cast<u32>(Channels::FrontRight)] = + static_cast<s16>(std::clamp(right_sample, min, max)); + } + + samples.resize(samples.size() / system_channels * device_channels); + + } else if (system_channels == 2 && device_channels == 6) { + // We need moar samples! Not all games will provide 6 channel audio. + // TODO: Implement some upmixing here. Currently just passthrough, with other + // channels left as silence. + std::vector<s16> new_samples(samples.size() / system_channels * device_channels, 0); + + for (u32 read_index = 0, write_index = 0; read_index < samples.size(); + read_index += system_channels, write_index += device_channels) { + const auto left_sample{static_cast<s16>(std::clamp( + static_cast<s32>( + static_cast<f32>( + samples[read_index + static_cast<u32>(Channels::FrontLeft)]) * + volume), + min, max))}; + + new_samples[write_index + static_cast<u32>(Channels::FrontLeft)] = left_sample; + + const auto right_sample{static_cast<s16>(std::clamp( + static_cast<s32>( + static_cast<f32>( + samples[read_index + static_cast<u32>(Channels::FrontRight)]) * + volume), + min, max))}; + + new_samples[write_index + static_cast<u32>(Channels::FrontRight)] = + right_sample; + } + samples = std::move(new_samples); + + } else if (volume != 1.0f) { + for (u32 i = 0; i < samples.size(); i++) { + samples[i] = static_cast<s16>(std::clamp( + static_cast<s32>(static_cast<f32>(samples[i]) * volume), min, max)); + } + } + + samples_buffer.Push(samples); + queue.enqueue(buffer); + queued_buffers++; + } + } + + /** + * Release a buffer. Audio In only, will fill a buffer with recorded samples. + * + * @param num_samples - Maximum number of samples to receive. + * @return Vector of recorded samples. May have fewer than num_samples. + */ + std::vector<s16> ReleaseBuffer(const u64 num_samples) override { + static constexpr s32 min = std::numeric_limits<s16>::min(); + static constexpr s32 max = std::numeric_limits<s16>::max(); + + auto samples{samples_buffer.Pop(num_samples)}; + + // TODO: Up-mix to 6 channels if the game expects it. + // For audio input this is unlikely to ever be the case though. + + // Incoming mic volume seems to always be very quiet, so multiply by an additional 8 here. + // TODO: Play with this and find something that works better. + auto volume{system_volume * device_volume * 8}; + for (u32 i = 0; i < samples.size(); i++) { + samples[i] = static_cast<s16>( + std::clamp(static_cast<s32>(static_cast<f32>(samples[i]) * volume), min, max)); + } + + if (samples.size() < num_samples) { + samples.resize(num_samples, 0); + } + return samples; + } + + /** + * Check if a certain buffer has been consumed (fully played). + * + * @param tag - Unique tag of a buffer to check for. + * @return True if the buffer has been played, otherwise false. + */ + bool IsBufferConsumed(const u64 tag) override { + if (released_buffer.tag == 0) { + if (!released_buffers.try_dequeue(released_buffer)) { + return false; + } + } + + if (released_buffer.tag == tag) { + released_buffer.tag = 0; + return true; + } + return false; + } + + /** + * Empty out the buffer queue. + */ + void ClearQueue() override { + samples_buffer.Pop(); + while (queue.pop()) { + } + while (released_buffers.pop()) { + } + released_buffer = {}; + playing_buffer = {}; + playing_buffer.consumed = true; + queued_buffers = 0; + } + +private: + /** + * Signal events back to the audio system that a buffer was played/can be filled. + * + * @param buffer - Consumed audio buffer to be released. + */ + void SignalEvent(const ::AudioCore::Sink::SinkBuffer& buffer) { + auto& manager{system.AudioCore().GetAudioManager()}; + switch (type) { + case StreamType::Out: + released_buffers.enqueue(buffer); + manager.SetEvent(Event::Type::AudioOutManager, true); + break; + case StreamType::In: + released_buffers.enqueue(buffer); + manager.SetEvent(Event::Type::AudioInManager, true); + break; + case StreamType::Render: + break; + } + } + + /** + * Main callback from SDL. Either expects samples from us (audio render/audio out), or will + * provide samples to be copied (audio in). + * + * @param userdata - Custom data pointer passed along, points to a SDLSinkStream. + * @param stream - Buffer of samples to be filled or read. + * @param len - Length of the stream in bytes. + */ + static void DataCallback(void* userdata, Uint8* stream, int len) { + auto* impl = static_cast<SDLSinkStream*>(userdata); + + if (!impl) { + return; + } + + const std::size_t num_channels = impl->GetDeviceChannels(); + const std::size_t frame_size = num_channels; + const std::size_t frame_size_bytes = frame_size * sizeof(s16); + const std::size_t num_frames{len / num_channels / sizeof(s16)}; + size_t frames_written{0}; + [[maybe_unused]] bool underrun{false}; + + if (impl->type == StreamType::In) { + std::span<s16> input_buffer{reinterpret_cast<s16*>(stream), num_frames * frame_size}; + + while (frames_written < num_frames) { + auto& playing_buffer{impl->playing_buffer}; + + // If the playing buffer has been consumed or has no frames, we need a new one + if (playing_buffer.consumed || playing_buffer.frames == 0) { + if (!impl->queue.try_dequeue(impl->playing_buffer)) { + // If no buffer was available we've underrun, just push the samples and + // continue. + underrun = true; + impl->samples_buffer.Push(&input_buffer[frames_written * frame_size], + (num_frames - frames_written) * frame_size); + frames_written = num_frames; + continue; + } else { + impl->queued_buffers--; + impl->SignalEvent(impl->playing_buffer); + } + } + + // Get the minimum frames available between the currently playing buffer, and the + // amount we have left to fill + size_t frames_available{ + std::min(playing_buffer.frames - playing_buffer.frames_played, + num_frames - frames_written)}; + + impl->samples_buffer.Push(&input_buffer[frames_written * frame_size], + frames_available * frame_size); + + frames_written += frames_available; + playing_buffer.frames_played += frames_available; + + // If that's all the frames in the current buffer, add its samples and mark it as + // consumed + if (playing_buffer.frames_played >= playing_buffer.frames) { + impl->AddPlayedSampleCount(playing_buffer.frames_played * num_channels); + impl->playing_buffer.consumed = true; + } + } + + std::memcpy(&impl->last_frame[0], &input_buffer[(frames_written - 1) * frame_size], + frame_size_bytes); + } else { + std::span<s16> output_buffer{reinterpret_cast<s16*>(stream), num_frames * frame_size}; + + while (frames_written < num_frames) { + auto& playing_buffer{impl->playing_buffer}; + + // If the playing buffer has been consumed or has no frames, we need a new one + if (playing_buffer.consumed || playing_buffer.frames == 0) { + if (!impl->queue.try_dequeue(impl->playing_buffer)) { + // If no buffer was available we've underrun, fill the remaining buffer with + // the last written frame and continue. + underrun = true; + for (size_t i = frames_written; i < num_frames; i++) { + std::memcpy(&output_buffer[i * frame_size], &impl->last_frame[0], + frame_size_bytes); + } + frames_written = num_frames; + continue; + } else { + impl->queued_buffers--; + impl->SignalEvent(impl->playing_buffer); + } + } + + // Get the minimum frames available between the currently playing buffer, and the + // amount we have left to fill + size_t frames_available{ + std::min(playing_buffer.frames - playing_buffer.frames_played, + num_frames - frames_written)}; + + impl->samples_buffer.Pop(&output_buffer[frames_written * frame_size], + frames_available * frame_size); + + frames_written += frames_available; + playing_buffer.frames_played += frames_available; + + // If that's all the frames in the current buffer, add its samples and mark it as + // consumed + if (playing_buffer.frames_played >= playing_buffer.frames) { + impl->AddPlayedSampleCount(playing_buffer.frames_played * num_channels); + impl->playing_buffer.consumed = true; + } + } + + std::memcpy(&impl->last_frame[0], &output_buffer[(frames_written - 1) * frame_size], + frame_size_bytes); + } + } + + /// SDL device id of the opened input/output device + SDL_AudioDeviceID device{}; + /// Type of this stream + StreamType type; + /// Core system + Core::System& system; + /// Ring buffer of the samples waiting to be played or consumed + Common::RingBuffer<s16, 0x10000> samples_buffer; + /// Audio buffers queued and waiting to play + Common::ReaderWriterQueue<::AudioCore::Sink::SinkBuffer> queue; + /// The currently-playing audio buffer + ::AudioCore::Sink::SinkBuffer playing_buffer{}; + /// Audio buffers which have been played and are in queue to be released by the audio system + Common::ReaderWriterQueue<::AudioCore::Sink::SinkBuffer> released_buffers{}; + /// Currently released buffer waiting to be taken by the audio system + ::AudioCore::Sink::SinkBuffer released_buffer{}; + /// The last played (or received) frame of audio, used when the callback underruns + std::array<s16, MaxChannels> last_frame{}; +}; + +SDLSink::SDLSink(std::string_view target_device_name) { + if (!SDL_WasInit(SDL_INIT_AUDIO)) { + if (SDL_InitSubSystem(SDL_INIT_AUDIO) < 0) { + LOG_CRITICAL(Audio_Sink, "SDL_InitSubSystem audio failed: {}", SDL_GetError()); + return; + } + } + + if (target_device_name != auto_device_name && !target_device_name.empty()) { + output_device = target_device_name; + } else { + output_device.clear(); + } + + device_channels = 2; +} + +SDLSink::~SDLSink() = default; + +SinkStream* SDLSink::AcquireSinkStream(Core::System& system, const u32 system_channels, + const std::string&, const StreamType type) { + SinkStreamPtr& stream = sink_streams.emplace_back(std::make_unique<SDLSinkStream>( + device_channels, system_channels, output_device, input_device, type, system)); + return stream.get(); +} + +void SDLSink::CloseStream(const SinkStream* stream) { + for (size_t i = 0; i < sink_streams.size(); i++) { + if (sink_streams[i].get() == stream) { + sink_streams[i].reset(); + sink_streams.erase(sink_streams.begin() + i); + break; + } + } +} + +void SDLSink::CloseStreams() { + sink_streams.clear(); +} + +void SDLSink::PauseStreams() { + for (auto& stream : sink_streams) { + stream->Stop(); + } +} + +void SDLSink::UnpauseStreams() { + for (auto& stream : sink_streams) { + stream->Start(); + } +} + +f32 SDLSink::GetDeviceVolume() const { + if (sink_streams.empty()) { + return 1.0f; + } + + return sink_streams[0]->GetDeviceVolume(); +} + +void SDLSink::SetDeviceVolume(const f32 volume) { + for (auto& stream : sink_streams) { + stream->SetDeviceVolume(volume); + } +} + +void SDLSink::SetSystemVolume(const f32 volume) { + for (auto& stream : sink_streams) { + stream->SetSystemVolume(volume); + } +} + +std::vector<std::string> ListSDLSinkDevices(const bool capture) { + std::vector<std::string> device_list; + + if (!SDL_WasInit(SDL_INIT_AUDIO)) { + if (SDL_InitSubSystem(SDL_INIT_AUDIO) < 0) { + LOG_CRITICAL(Audio_Sink, "SDL_InitSubSystem audio failed: {}", SDL_GetError()); + return {}; + } + } + + const int device_count = SDL_GetNumAudioDevices(capture); + for (int i = 0; i < device_count; ++i) { + device_list.emplace_back(SDL_GetAudioDeviceName(i, 0)); + } + + return device_list; +} + +} // namespace AudioCore::Sink diff --git a/src/audio_core/sink/sdl2_sink.h b/src/audio_core/sink/sdl2_sink.h new file mode 100644 index 000000000..186bc2fa3 --- /dev/null +++ b/src/audio_core/sink/sdl2_sink.h @@ -0,0 +1,101 @@ +// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#pragma once + +#include <string> +#include <vector> + +#include "audio_core/sink/sink.h" + +namespace Core { +class System; +} + +namespace AudioCore::Sink { +class SinkStream; + +/** + * SDL backend sink, holds multiple output streams and is responsible for sinking samples to + * hardware. Used by Audio Render, Audio In and Audio Out. + */ +class SDLSink final : public Sink { +public: + explicit SDLSink(std::string_view device_id); + ~SDLSink() override; + + /** + * Create a new sink stream. + * + * @param system - Core system. + * @param system_channels - Number of channels the audio system expects. + * May differ from the device's channel count. + * @param name - Name of this stream. + * @param type - Type of this stream, render/in/out. + * @param event - Audio render only, a signal used to prevent the renderer running too + * fast. + * @return A pointer to the created SinkStream + */ + SinkStream* AcquireSinkStream(Core::System& system, u32 system_channels, + const std::string& name, StreamType type) override; + + /** + * Close a given stream. + * + * @param stream - The stream to close. + */ + void CloseStream(const SinkStream* stream) override; + + /** + * Close all streams. + */ + void CloseStreams() override; + + /** + * Pause all streams. + */ + void PauseStreams() override; + + /** + * Unpause all streams. + */ + void UnpauseStreams() override; + + /** + * Get the device volume. Set from calls to the IAudioDevice service. + * + * @return Volume of the device. + */ + f32 GetDeviceVolume() const override; + + /** + * Set the device volume. Set from calls to the IAudioDevice service. + * + * @param volume - New volume of the device. + */ + void SetDeviceVolume(f32 volume) override; + + /** + * Set the system volume. Comes from the audio system using this stream. + * + * @param volume - New volume of the system. + */ + void SetSystemVolume(f32 volume) override; + +private: + /// Name of the output device used by streams + std::string output_device; + /// Name of the input device used by streams + std::string input_device; + /// Vector of streams managed by this sink + std::vector<SinkStreamPtr> sink_streams; +}; + +/** + * Get a list of conencted devices from Cubeb. + * + * @param capture - Return input (capture) devices if true, otherwise output devices. + */ +std::vector<std::string> ListSDLSinkDevices(bool capture); + +} // namespace AudioCore::Sink diff --git a/src/audio_core/sink/sink.h b/src/audio_core/sink/sink.h new file mode 100644 index 000000000..91fe455e4 --- /dev/null +++ b/src/audio_core/sink/sink.h @@ -0,0 +1,106 @@ +// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#pragma once + +#include <memory> +#include <string> + +#include "audio_core/sink/sink_stream.h" +#include "common/common_types.h" + +namespace Common { +class Event; +} +namespace Core { +class System; +} + +namespace AudioCore::Sink { + +constexpr char auto_device_name[] = "auto"; + +/** + * This class is an interface for an audio sink, holds multiple output streams and is responsible + * for sinking samples to hardware. Used by Audio Render, Audio In and Audio Out. + */ +class Sink { +public: + virtual ~Sink() = default; + /** + * Close a given stream. + * + * @param stream - The stream to close. + */ + virtual void CloseStream(const SinkStream* stream) = 0; + + /** + * Close all streams. + */ + virtual void CloseStreams() = 0; + + /** + * Pause all streams. + */ + virtual void PauseStreams() = 0; + + /** + * Unpause all streams. + */ + virtual void UnpauseStreams() = 0; + + /** + * Create a new sink stream, kept within this sink, with a pointer returned for use. + * Do not free the returned pointer. When done with the stream, call CloseStream on the sink. + * + * @param system - Core system. + * @param system_channels - Number of channels the audio system expects. + * May differ from the device's channel count. + * @param name - Name of this stream. + * @param type - Type of this stream, render/in/out. + * @param event - Audio render only, a signal used to prevent the renderer running too + * fast. + * @return A pointer to the created SinkStream + */ + virtual SinkStream* AcquireSinkStream(Core::System& system, u32 system_channels, + const std::string& name, StreamType type) = 0; + + /** + * Get the number of channels the hardware device supports. + * Either 2 or 6. + * + * @return Number of device channels. + */ + u32 GetDeviceChannels() const { + return device_channels; + } + + /** + * Get the device volume. Set from calls to the IAudioDevice service. + * + * @return Volume of the device. + */ + virtual f32 GetDeviceVolume() const = 0; + + /** + * Set the device volume. Set from calls to the IAudioDevice service. + * + * @param volume - New volume of the device. + */ + virtual void SetDeviceVolume(f32 volume) = 0; + + /** + * Set the system volume. Comes from the audio system using this stream. + * + * @param volume - New volume of the system. + */ + virtual void SetSystemVolume(f32 volume) = 0; + +protected: + /// Number of device channels supported by the hardware + u32 device_channels{2}; +}; + +using SinkPtr = std::unique_ptr<Sink>; + +} // namespace AudioCore::Sink diff --git a/src/audio_core/sink/sink_details.cpp b/src/audio_core/sink/sink_details.cpp new file mode 100644 index 000000000..253c0fd1e --- /dev/null +++ b/src/audio_core/sink/sink_details.cpp @@ -0,0 +1,91 @@ +// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#include <algorithm> +#include <memory> +#include <string> +#include <vector> +#include "audio_core/sink/null_sink.h" +#include "audio_core/sink/sink_details.h" +#ifdef HAVE_CUBEB +#include "audio_core/sink/cubeb_sink.h" +#endif +#ifdef HAVE_SDL2 +#include "audio_core/sink/sdl2_sink.h" +#endif +#include "common/logging/log.h" + +namespace AudioCore::Sink { +namespace { +struct SinkDetails { + using FactoryFn = std::unique_ptr<Sink> (*)(std::string_view); + using ListDevicesFn = std::vector<std::string> (*)(bool); + + /// Name for this sink. + const char* id; + /// A method to call to construct an instance of this type of sink. + FactoryFn factory; + /// A method to call to list available devices. + ListDevicesFn list_devices; +}; + +// sink_details is ordered in terms of desirability, with the best choice at the top. +constexpr SinkDetails sink_details[] = { +#ifdef HAVE_CUBEB + SinkDetails{"cubeb", + [](std::string_view device_id) -> std::unique_ptr<Sink> { + return std::make_unique<CubebSink>(device_id); + }, + &ListCubebSinkDevices}, +#endif +#ifdef HAVE_SDL2 + SinkDetails{"sdl2", + [](std::string_view device_id) -> std::unique_ptr<Sink> { + return std::make_unique<SDLSink>(device_id); + }, + &ListSDLSinkDevices}, +#endif + SinkDetails{"null", + [](std::string_view device_id) -> std::unique_ptr<Sink> { + return std::make_unique<NullSink>(device_id); + }, + [](bool capture) { return std::vector<std::string>{"null"}; }}, +}; + +const SinkDetails& GetOutputSinkDetails(std::string_view sink_id) { + auto iter = + std::find_if(std::begin(sink_details), std::end(sink_details), + [sink_id](const auto& sink_detail) { return sink_detail.id == sink_id; }); + + if (sink_id == "auto" || iter == std::end(sink_details)) { + if (sink_id != "auto") { + LOG_ERROR(Audio, "AudioCore::Sink::GetOutputSinkDetails given invalid sink_id {}", + sink_id); + } + // Auto-select. + // sink_details is ordered in terms of desirability, with the best choice at the front. + iter = std::begin(sink_details); + } + + return *iter; +} +} // Anonymous namespace + +std::vector<const char*> GetSinkIDs() { + std::vector<const char*> sink_ids(std::size(sink_details)); + + std::transform(std::begin(sink_details), std::end(sink_details), std::begin(sink_ids), + [](const auto& sink) { return sink.id; }); + + return sink_ids; +} + +std::vector<std::string> GetDeviceListForSink(std::string_view sink_id, bool capture) { + return GetOutputSinkDetails(sink_id).list_devices(capture); +} + +std::unique_ptr<Sink> CreateSinkFromID(std::string_view sink_id, std::string_view device_id) { + return GetOutputSinkDetails(sink_id).factory(device_id); +} + +} // namespace AudioCore::Sink diff --git a/src/audio_core/sink/sink_details.h b/src/audio_core/sink/sink_details.h new file mode 100644 index 000000000..3ebdb1e30 --- /dev/null +++ b/src/audio_core/sink/sink_details.h @@ -0,0 +1,43 @@ +// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#pragma once + +#include <string> +#include <string_view> +#include <vector> + +namespace AudioCore { +class AudioManager; + +namespace Sink { + +class Sink; + +/** + * Retrieves the IDs for all available audio sinks. + * + * @return Vector of available sink names. + */ +std::vector<const char*> GetSinkIDs(); + +/** + * Gets the list of devices for a particular sink identified by the given ID. + * + * @param sink_id - Id of the sink to get devices from. + * @param capture - Get capture (input) devices, or output devices? + * @return Vector of device names. + */ +std::vector<std::string> GetDeviceListForSink(std::string_view sink_id, bool capture); + +/** + * Creates an audio sink identified by the given device ID. + * + * @param sink_id - Id of the sink to create. + * @param device_id - Name of the device to create. + * @return Pointer to the created sink. + */ +std::unique_ptr<Sink> CreateSinkFromID(std::string_view sink_id, std::string_view device_id); + +} // namespace Sink +} // namespace AudioCore diff --git a/src/audio_core/sink/sink_stream.h b/src/audio_core/sink/sink_stream.h new file mode 100644 index 000000000..17ed6593f --- /dev/null +++ b/src/audio_core/sink/sink_stream.h @@ -0,0 +1,224 @@ +// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#pragma once + +#include <atomic> +#include <memory> +#include <vector> + +#include "audio_core/common/common.h" +#include "common/common_types.h" + +namespace AudioCore::Sink { + +enum class StreamType { + Render, + Out, + In, +}; + +struct SinkBuffer { + u64 frames; + u64 frames_played; + u64 tag; + bool consumed; +}; + +/** + * Contains a real backend stream for outputting samples to hardware, + * created only via a Sink (See Sink::AcquireSinkStream). + * + * Accepts a SinkBuffer and samples in PCM16 format to be output (see AppendBuffer). + * Appended buffers act as a FIFO queue, and will be held until played. + * You should regularly call IsBufferConsumed with the unique SinkBuffer tag to check if the buffer + * has been consumed. + * + * Since these are a FIFO queue, always check IsBufferConsumed in the same order you appended the + * buffers, skipping a buffer will result in all following buffers to never release. + * + * If the buffers appear to be stuck, you can stop and re-open an IAudioIn/IAudioOut service (this + * is what games do), or call ClearQueue to flush all of the buffers without a full restart. + */ +class SinkStream { +public: + virtual ~SinkStream() = default; + + /** + * Finalize the sink stream. + */ + virtual void Finalize() = 0; + + /** + * Start the sink stream. + * + * @param resume - Set to true if this is resuming the stream a previously-active stream. + * Default false. + */ + virtual void Start(bool resume = false) = 0; + + /** + * Stop the sink stream. + */ + virtual void Stop() = 0; + + /** + * Append a new buffer and its samples to a waiting queue to play. + * + * @param buffer - Audio buffer information to be queued. + * @param samples - The s16 samples to be queue for playback. + */ + virtual void AppendBuffer(SinkBuffer& buffer, std::vector<s16>& samples) = 0; + + /** + * Release a buffer. Audio In only, will fill a buffer with recorded samples. + * + * @param num_samples - Maximum number of samples to receive. + * @return Vector of recorded samples. May have fewer than num_samples. + */ + virtual std::vector<s16> ReleaseBuffer(u64 num_samples) = 0; + + /** + * Check if a certain buffer has been consumed (fully played). + * + * @param tag - Unique tag of a buffer to check for. + * @return True if the buffer has been played, otherwise false. + */ + virtual bool IsBufferConsumed(u64 tag) = 0; + + /** + * Empty out the buffer queue. + */ + virtual void ClearQueue() = 0; + + /** + * Check if the stream is paused. + * + * @return True if paused, otherwise false. + */ + bool IsPaused() { + return paused; + } + + /** + * Get the number of system channels in this stream. + * + * @return Number of system channels. + */ + u32 GetSystemChannels() const { + return system_channels; + } + + /** + * Set the number of channels the system expects. + * + * @param channels - New number of system channels. + */ + void SetSystemChannels(u32 channels) { + system_channels = channels; + } + + /** + * Get the number of channels the hardware supports. + * + * @return Number of channels supported. + */ + u32 GetDeviceChannels() const { + return device_channels; + } + + /** + * Get the total number of samples played by this stream. + * + * @return Number of samples played. + */ + u64 GetPlayedSampleCount() const { + return played_sample_count; + } + + /** + * Set the number of samples played. + * This is started and stopped on system start/stop. + * + * @param played_sample_count_ - Number of samples to set. + */ + void SetPlayedSampleCount(u64 played_sample_count_) { + played_sample_count = played_sample_count_; + } + + /** + * Add to the played sample count. + * + * @param num_samples - Number of samples to add. + */ + void AddPlayedSampleCount(u64 num_samples) { + played_sample_count += num_samples; + } + + /** + * Get the system volume. + * + * @return The current system volume. + */ + f32 GetSystemVolume() const { + return system_volume; + } + + /** + * Get the device volume. + * + * @return The current device volume. + */ + f32 GetDeviceVolume() const { + return device_volume; + } + + /** + * Set the system volume. + * + * @param volume_ - The new system volume. + */ + void SetSystemVolume(f32 volume_) { + system_volume = volume_; + } + + /** + * Set the device volume. + * + * @param volume_ - The new device volume. + */ + void SetDeviceVolume(f32 volume_) { + device_volume = volume_; + } + + /** + * Get the number of queued audio buffers. + * + * @return The number of queued buffers. + */ + u32 GetQueueSize() { + return queued_buffers.load(); + } + +protected: + /// Number of buffers waiting to be played + std::atomic<u32> queued_buffers{}; + /// Total samples played by this stream + std::atomic<u64> played_sample_count{}; + /// Set by the audio render/in/out system which uses this stream + f32 system_volume{1.0f}; + /// Set via IAudioDevice service calls + f32 device_volume{1.0f}; + /// Set by the audio render/in/out systen which uses this stream + u32 system_channels{2}; + /// Channels supported by hardware + u32 device_channels{2}; + /// Is this stream currently paused? + std::atomic<bool> paused{true}; + /// Was this stream previously playing? + std::atomic<bool> was_playing{false}; +}; + +using SinkStreamPtr = std::unique_ptr<SinkStream>; + +} // namespace AudioCore::Sink |