#include #include #include #include #include #include #include #include #include "../../ultramodern/ultra64.h" #include "../../ultramodern/ultramodern.hpp" #define SDL_MAIN_HANDLED #ifdef _WIN32 #include "SDL.h" #else #include "SDL2/SDL.h" #include "SDL2/SDL_syswm.h" #endif #include "recomp_ui.h" #include "recomp_input.h" #include "recomp_config.h" #ifdef _WIN32 #define WIN32_LEAN_AND_MEAN #include #include "SDL_syswm.h" #endif extern "C" void init(); /*extern "C"*/ void start(ultramodern::WindowHandle window_handle, const ultramodern::audio_callbacks_t* audio_callbacks, const ultramodern::input_callbacks_t* input_callbacks); template void exit_error(const char* str, Ts ...args) { // TODO pop up an error ((void)fprintf(stderr, str, args), ...); assert(false); std::quick_exit(EXIT_FAILURE); } ultramodern::gfx_callbacks_t::gfx_data_t create_gfx() { SDL_SetHint(SDL_HINT_WINDOWS_DPI_AWARENESS, "system"); SDL_SetHint(SDL_HINT_GAMECONTROLLER_USE_BUTTON_LABELS, "0"); if (SDL_Init(SDL_INIT_VIDEO | SDL_INIT_GAMECONTROLLER) > 0) { exit_error("Failed to initialize SDL2: %s\n", SDL_GetError()); } return {}; } SDL_Window* window; ultramodern::WindowHandle create_window(ultramodern::gfx_callbacks_t::gfx_data_t) { window = SDL_CreateWindow("Zelda 64: Recompiled", SDL_WINDOWPOS_CENTERED, SDL_WINDOWPOS_CENTERED, 1280, 720, SDL_WINDOW_RESIZABLE ); if (window == nullptr) { exit_error("Failed to create window: %s\n", SDL_GetError()); } SDL_SysWMinfo wmInfo; SDL_VERSION(&wmInfo.version); SDL_GetWindowWMInfo(window, &wmInfo); #if defined(_WIN32) return ultramodern::WindowHandle{ wmInfo.info.win.window, GetCurrentThreadId() }; #elif defined(__ANDROID__) static_assert(false && "Unimplemented"); #elif defined(__linux__) return ultramodern::WindowHandle{ wmInfo.info.x11.display, wmInfo.info.x11.window }; #else static_assert(false && "Unimplemented"); #endif } void update_gfx(void*) { recomp::handle_events(); } static SDL_AudioCVT audio_convert; static SDL_AudioDeviceID audio_device = 0; // Samples per channel per second. static uint32_t sample_rate = 48000; static uint32_t output_sample_rate = 48000; // Channel count. constexpr uint32_t input_channels = 2; static uint32_t output_channels = 2; // Terminology: a frame is a collection of samples for each channel. e.g. 2 input samples is one input frame. This is unrelated to graphical frames. // Number of frames to duplicate for fixing interpolation at the start and end of a chunk. constexpr uint32_t duplicated_input_frames = 4; // The number of output frames to skip for playback (to avoid playing duplicate inputs twice). static uint32_t discarded_output_frames; void queue_samples(int16_t* audio_data, size_t sample_count) { // Buffer for holding the output of swapping the audio channels. This is reused across // calls to reduce runtime allocations. static std::vector swap_buffer; static std::array duplicated_sample_buffer; // Make sure the swap buffer is large enough to hold the audio data, including any extra space needed for resampling. size_t resampled_sample_count = sample_count + duplicated_input_frames * input_channels; size_t max_sample_count = std::max(resampled_sample_count, resampled_sample_count * audio_convert.len_mult); if (max_sample_count > swap_buffer.size()) { swap_buffer.resize(max_sample_count); } // Copy the duplicated frames from last chunk into this chunk for (size_t i = 0; i < duplicated_input_frames * input_channels; i++) { swap_buffer[i] = duplicated_sample_buffer[i]; } // Convert the audio from 16-bit values to floats and swap the audio channels into the // swap buffer to correct for the address xor caused by endianness handling. for (size_t i = 0; i < sample_count; i += input_channels) { swap_buffer[i + 0 + duplicated_input_frames * input_channels] = audio_data[i + 1] * (0.5f / 32768.0f); swap_buffer[i + 1 + duplicated_input_frames * input_channels] = audio_data[i + 0] * (0.5f / 32768.0f); } // TODO handle cases where a chunk is smaller than the duplicated frame count. assert(sample_count > duplicated_input_frames * input_channels); // Copy the last converted samples into the duplicated sample buffer to reuse in resampling the next queued chunk. for (size_t i = 0; i < duplicated_input_frames * input_channels; i++) { duplicated_sample_buffer[i] = swap_buffer[i + sample_count]; } audio_convert.buf = reinterpret_cast(swap_buffer.data()); audio_convert.len = (sample_count + duplicated_input_frames * input_channels) * sizeof(swap_buffer[0]); int ret = SDL_ConvertAudio(&audio_convert); if (ret < 0) { printf("Error using SDL audio converter: %s\n", SDL_GetError()); throw std::runtime_error("Error using SDL audio converter"); } // Queue the swapped audio data. // Offset the data start by only half the discarded frame count as the other half of the discarded frames are at the end of the buffer. SDL_QueueAudio(audio_device, swap_buffer.data() + output_channels * discarded_output_frames / 2, audio_convert.len_cvt - output_channels * discarded_output_frames * sizeof(swap_buffer[0])); } constexpr uint32_t bytes_per_frame = input_channels * sizeof(float); size_t get_frames_remaining() { constexpr float buffer_offset_frames = 1.0f; // Get the number of remaining buffered audio bytes. uint32_t buffered_byte_count = SDL_GetQueuedAudioSize(audio_device); // Scale the byte count based on the ratio of sample rates and channel counts. buffered_byte_count = buffered_byte_count * 2 * sample_rate / output_sample_rate / output_channels; // Adjust the reported count to be some number of refreshes in the future, which helps ensure that // there are enough samples even if the audio thread experiences a small amount of lag. This prevents // audio popping on games that use the buffered audio byte count to determine how many samples // to generate. uint32_t frames_per_vi = (sample_rate / 60); if (buffered_byte_count > (buffer_offset_frames * bytes_per_frame * frames_per_vi)) { buffered_byte_count -= (buffer_offset_frames * bytes_per_frame * frames_per_vi); } else { buffered_byte_count = 0; } // Convert from byte count to sample count. return buffered_byte_count / bytes_per_frame; } void update_audio_converter() { int ret = SDL_BuildAudioCVT(&audio_convert, AUDIO_F32, input_channels, sample_rate, AUDIO_F32, output_channels, output_sample_rate); if (ret < 0) { printf("Error creating SDL audio converter: %s\n", SDL_GetError()); throw std::runtime_error("Error creating SDL audio converter"); } // Calculate the number of samples to discard based on the sample rate ratio and the duplicate frame count. discarded_output_frames = duplicated_input_frames * output_sample_rate / sample_rate; } void set_frequency(uint32_t freq) { sample_rate = freq; update_audio_converter(); } void reset_audio(uint32_t output_freq) { SDL_AudioSpec spec_desired{ .freq = (int)output_freq, .format = AUDIO_F32, .channels = (Uint8)output_channels, .silence = 0, // calculated .samples = 0x100, // Fairly small sample count to reduce the latency of internal buffering .padding = 0, // unused .size = 0, // calculated .callback = nullptr, .userdata = nullptr }; audio_device = SDL_OpenAudioDevice(nullptr, false, &spec_desired, nullptr, 0); if (audio_device == 0) { exit_error("SDL error opening audio device: %s\n", SDL_GetError()); } SDL_PauseAudioDevice(audio_device, 0); output_sample_rate = output_freq; update_audio_converter(); } int main(int argc, char** argv) { #ifdef _WIN32 // Set up console output to accept UTF-8 on windows SetConsoleOutputCP(CP_UTF8); // Change to a font that supports Japanese characters CONSOLE_FONT_INFOEX cfi; cfi.cbSize = sizeof cfi; cfi.nFont = 0; cfi.dwFontSize.X = 0; cfi.dwFontSize.Y = 16; cfi.FontFamily = FF_DONTCARE; cfi.FontWeight = FW_NORMAL; wcscpy_s(cfi.FaceName, L"NSimSun"); SetCurrentConsoleFontEx(GetStdHandle(STD_OUTPUT_HANDLE), FALSE, &cfi); #else std::setlocale(LC_ALL, "en_US.UTF-8"); #endif printf("Current dir: %ls\n", std::filesystem::current_path().c_str()); // Initialize SDL audio and set the output frequency. SDL_InitSubSystem(SDL_INIT_AUDIO); reset_audio(48000); init(); recomp::load_config(); ultramodern::gfx_callbacks_t gfx_callbacks{ .create_gfx = create_gfx, .create_window = create_window, .update_gfx = update_gfx, }; ultramodern::audio_callbacks_t audio_callbacks{ .queue_samples = queue_samples, .get_frames_remaining = get_frames_remaining, .set_frequency = set_frequency, }; ultramodern::input_callbacks_t input_callbacks{ .poll_input = recomp::poll_inputs, .get_input = recomp::get_n64_input, }; ultramodern::start({}, audio_callbacks, input_callbacks, gfx_callbacks); return EXIT_SUCCESS; }