diff options
Diffstat (limited to 'src/audio_core/sink/sdl2_sink.cpp')
| -rw-r--r-- | src/audio_core/sink/sdl2_sink.cpp | 377 |
1 files changed, 32 insertions, 345 deletions
diff --git a/src/audio_core/sink/sdl2_sink.cpp b/src/audio_core/sink/sdl2_sink.cpp index d6c9ec90d..1bd001b94 100644 --- a/src/audio_core/sink/sdl2_sink.cpp +++ b/src/audio_core/sink/sdl2_sink.cpp @@ -1,20 +1,13 @@ // SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project // SPDX-License-Identifier: GPL-2.0-or-later -#include <algorithm> -#include <atomic> +#include <span> +#include <vector> -#include "audio_core/audio_core.h" -#include "audio_core/audio_event.h" -#include "audio_core/audio_manager.h" +#include "audio_core/common/common.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 @@ -44,10 +37,9 @@ public: * @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_} { + SDLSinkStream(u32 device_channels_, u32 system_channels_, const std::string& output_device, + const std::string& input_device, StreamType type_, Core::System& system_) + : SinkStream{system_, type_} { system_channels = system_channels_; device_channels = device_channels_; @@ -63,8 +55,6 @@ public: spec.callback = &SDLSinkStream::DataCallback; spec.userdata = this; - playing_buffer.consumed = true; - std::string device_name{output_device}; bool capture{false}; if (type == StreamType::In) { @@ -84,31 +74,30 @@ public: return; } - LOG_DEBUG(Service_Audio, - "Opening sdl stream {} with: rate {} channels {} (system channels {}) " - " samples {}", - device, obtained.freq, obtained.channels, system_channels, obtained.samples); + LOG_INFO(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); + LOG_DEBUG(Service_Audio, "Destructing SDL stream {}", name); + Finalize(); } /** * Finalize the sink stream. */ void Finalize() override { + Unstall(); if (device == 0) { return; } + Stop(); SDL_CloseAudioDevice(device); } @@ -118,217 +107,29 @@ public: * @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) { + void Start(bool resume = false) override { + if (device == 0 || !paused) { return; } - if (resume && was_playing) { - SDL_PauseAudioDevice(device, 0); - paused = false; - } else if (!resume) { - SDL_PauseAudioDevice(device, 0); - paused = false; - } + paused = false; + SDL_PauseAudioDevice(device, 0); } /** * Stop the sink stream. */ - void Stop() { - if (device == 0) { + void Stop() override { + Unstall(); + if (device == 0 || paused) { 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; + SDL_PauseAudioDevice(device, 1); } 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). * @@ -345,122 +146,20 @@ private: 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); + std::span<const s16> input_buffer{reinterpret_cast<const s16*>(stream), + num_frames * frame_size}; + impl->ProcessAudioIn(input_buffer, num_frames); } 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); + impl->ProcessAudioOutAndRender(output_buffer, num_frames); } } /// 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) { @@ -482,14 +181,14 @@ SDLSink::SDLSink(std::string_view target_device_name) { SDLSink::~SDLSink() = default; -SinkStream* SDLSink::AcquireSinkStream(Core::System& system, const u32 system_channels, - const std::string&, const StreamType type) { +SinkStream* SDLSink::AcquireSinkStream(Core::System& system, u32 system_channels, + const std::string&, 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) { +void SDLSink::CloseStream(SinkStream* stream) { for (size_t i = 0; i < sink_streams.size(); i++) { if (sink_streams[i].get() == stream) { sink_streams[i].reset(); @@ -503,18 +202,6 @@ 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; @@ -523,19 +210,19 @@ f32 SDLSink::GetDeviceVolume() const { return sink_streams[0]->GetDeviceVolume(); } -void SDLSink::SetDeviceVolume(const f32 volume) { +void SDLSink::SetDeviceVolume(f32 volume) { for (auto& stream : sink_streams) { stream->SetDeviceVolume(volume); } } -void SDLSink::SetSystemVolume(const f32 volume) { +void SDLSink::SetSystemVolume(f32 volume) { for (auto& stream : sink_streams) { stream->SetSystemVolume(volume); } } -std::vector<std::string> ListSDLSinkDevices(const bool capture) { +std::vector<std::string> ListSDLSinkDevices(bool capture) { std::vector<std::string> device_list; if (!SDL_WasInit(SDL_INIT_AUDIO)) { |