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Diffstat (limited to 'src/audio_core/sink/sink_stream.cpp')
-rw-r--r-- | src/audio_core/sink/sink_stream.cpp | 265 |
1 files changed, 265 insertions, 0 deletions
diff --git a/src/audio_core/sink/sink_stream.cpp b/src/audio_core/sink/sink_stream.cpp new file mode 100644 index 000000000..24636e512 --- /dev/null +++ b/src/audio_core/sink/sink_stream.cpp @@ -0,0 +1,265 @@ +// SPDX-FileCopyrightText: Copyright 2018 yuzu Emulator Project +// SPDX-License-Identifier: GPL-2.0-or-later + +#pragma once + +#include <array> +#include <atomic> +#include <memory> +#include <span> +#include <vector> + +#include "audio_core/audio_core.h" +#include "audio_core/common/common.h" +#include "audio_core/sink/sink_stream.h" +#include "common/common_types.h" +#include "common/fixed_point.h" +#include "common/settings.h" +#include "core/core.h" + +namespace AudioCore::Sink { + +void SinkStream::AppendBuffer(SinkBuffer& buffer, std::vector<s16>& samples) { + if (type == StreamType::In) { + queue.enqueue(buffer); + queued_buffers++; + return; + } + + constexpr s32 min{std::numeric_limits<s16>::min()}; + constexpr s32 max{std::numeric_limits<s16>::max()}; + + auto yuzu_volume{Settings::Volume()}; + if (yuzu_volume > 1.0f) { + yuzu_volume = 0.6f + 20 * std::log10(yuzu_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++; +} + +std::vector<s16> SinkStream::ReleaseBuffer(u64 num_samples) { + constexpr s32 min = std::numeric_limits<s16>::min(); + 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; +} + +void SinkStream::ClearQueue() { + samples_buffer.Pop(); + while (queue.pop()) { + } + queued_buffers = 0; + playing_buffer = {}; + playing_buffer.consumed = true; +} + +void SinkStream::ProcessAudioIn(std::span<const s16> input_buffer, std::size_t num_frames) { + const std::size_t num_channels = GetDeviceChannels(); + const std::size_t frame_size = num_channels; + const std::size_t frame_size_bytes = frame_size * sizeof(s16); + size_t frames_written{0}; + + if (queued_buffers > max_queue_size) { + Stall(); + } + + while (frames_written < num_frames) { + // 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 (!queue.try_dequeue(playing_buffer)) { + // If no buffer was available we've underrun, just push the samples and + // continue. + samples_buffer.Push(&input_buffer[frames_written * frame_size], + (num_frames - frames_written) * frame_size); + frames_written = num_frames; + continue; + } + // Successfully dequeued a new buffer. + queued_buffers--; + } + + // 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)}; + + 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) { + playing_buffer.consumed = true; + } + } + + std::memcpy(&last_frame[0], &input_buffer[(frames_written - 1) * frame_size], frame_size_bytes); + + if (queued_buffers <= max_queue_size) { + Unstall(); + } +} + +void SinkStream::ProcessAudioOutAndRender(std::span<s16> output_buffer, std::size_t num_frames) { + const std::size_t num_channels = GetDeviceChannels(); + const std::size_t frame_size = num_channels; + const std::size_t frame_size_bytes = frame_size * sizeof(s16); + size_t frames_written{0}; + + // Due to many frames being queued up with nvdec (5 frames or so?), a lot of buffers also get + // queued up (30+) but not all at once, which causes constant stalling here, so just let the + // video play out without attempting to stall. + // Can hopefully remove this later with a more complete NVDEC implementation. + const auto nvdec_active{system.AudioCore().IsNVDECActive()}; + if (!nvdec_active && queued_buffers > max_queue_size) { + Stall(); + } + + while (frames_written < num_frames) { + // 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 (!queue.try_dequeue(playing_buffer)) { + // If no buffer was available we've underrun, fill the remaining buffer with + // the last written frame and continue. + for (size_t i = frames_written; i < num_frames; i++) { + std::memcpy(&output_buffer[i * frame_size], &last_frame[0], frame_size_bytes); + } + frames_written = num_frames; + continue; + } + // Successfully dequeued a new buffer. + queued_buffers--; + } + + // 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)}; + + 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) { + playing_buffer.consumed = true; + } + } + + std::memcpy(&last_frame[0], &output_buffer[(frames_written - 1) * frame_size], + frame_size_bytes); + + if (stalled && queued_buffers <= max_queue_size) { + Unstall(); + } +} + +void SinkStream::Stall() { + if (stalled) { + return; + } + stalled = true; + system.StallProcesses(); +} + +void SinkStream::Unstall() { + if (!stalled) { + return; + } + system.UnstallProcesses(); + stalled = false; +} + +} // namespace AudioCore::Sink |