path: root/base/audio.c

#include <stdio.h>
#include <stdint.h>
#include <math.h>
#include <stdlib.h>

#define SAMPLE_RATE 48000
#define NUM_CHANNELS 2
#define FRAME_SIZE (NUM_CHANNELS * sizeof(short))

// static inline float smoothstep(float edge0, float edge1, float x) {
// 	x = (x - edge0) / (edge1 - edge0); // Scale x to [0, 1]
// 	x = x < 0.0f ? 0.0f : (x > 1.0f ? 1.0f : x); // Clamp to [0, 1]
// 	return x * x * (3.0f - 2.0f * x); // Smooth interpolation
// }

// static inline float smootherstep(float edge0, float edge1, float x) {
//     x = (x - edge0) / (edge1 - edge0); // Scale x to [0, 1]
//     x = x < 0.0f ? 0.0f : (x > 1.0f ? 1.0f : x); // Clamp to [0, 1]
//     return x * x * x * (x * (x * 6 - 15) + 10); // Modified curve
// }

static inline float fast_cos(float x) {
	float x2 = x * x;
	return 1.0f - x2 * (0.5f - x2 * 0.04166667f); // Approximation of cos(x)
}

static inline float cosine_smooth(float edge0, float edge1, float x) {
	x = (x - edge0) / (edge1 - edge0);				// Scale x to [0, 1]
	x = x < 0.0f ? 0.0f : (x > 1.0f ? 1.0f : x);	// Clamp to [0, 1]
	return 0.5f * (1.0f - fast_cos(x * M_PI));	// Cosine smoothing
}

static float filter_phase = 0.0f;
static float prev_output_sample_L = 0.0f;
static float prev_output_sample_R = 0.0f;

static void audio_callback_thread(int16_t *audio_buffer, size_t frames) {
	int filter_override = state.filter_override;										// Manual override: -1 = auto, 0 = off, 1 = on
	float filter_frequency = state.filter_frequency;									// Frequency in Hz for squarewave toggle

	audio_callback(audio_buffer, frames);

	if(filter_override) {
		float a = 1.0f * M_PI * 4000.0f / (SAMPLE_RATE + 1.0f * M_PI * 4000.0f);
		float phase_increment = filter_frequency / SAMPLE_RATE;

		for(size_t i = 0; i < frames * 2; i += 2) {
			float led_filter_active;

			if(filter_override == -1) {
				filter_phase += phase_increment;
				if(filter_phase >= 1.0f) filter_phase -= 1.0f;

				led_filter_active = cosine_smooth(0.45f, 0.50f, filter_phase) - cosine_smooth(0.95f, 1.00f, filter_phase);

			} else {
				led_filter_active = 1.0f;					// Manual override (1 = on)
			}

			float input_sample_L = (float)audio_buffer[i] / 32767.0f;
			float input_sample_R = (float)audio_buffer[i + 1] / 32767.0f;

			float filtered_sample_L = a * input_sample_L + (1.0f - a) * prev_output_sample_L;
			float filtered_sample_R = a * input_sample_R + (1.0f - a) * prev_output_sample_R;

			prev_output_sample_L = filtered_sample_L;
			prev_output_sample_R = filtered_sample_R;

			audio_buffer[i] = (int16_t)((1.0f - led_filter_active) * input_sample_L * 32767.0f + led_filter_active * filtered_sample_L * 32767.0f);
			audio_buffer[i + 1] = (int16_t)((1.0f - led_filter_active) * input_sample_R * 32767.0f + led_filter_active * filtered_sample_R * 32767.0f);
		}
	}
}

#ifdef __linux__


#include <pipewire/pipewire.h>
#include <spa/param/audio/format-utils.h>
#include <spa/param/props.h>

#define BUFFER_SIZE (512 * FRAME_SIZE)

static struct pw_thread_loop *pa_thread_loop;
static struct pw_context *pa_context;
static struct pw_core *pa_core;
static struct pw_stream *pa_stream;
static struct spa_hook pa_stream_listener;
static uint64_t audio_clock_frequency;
static uint64_t playback_cursor;

/*
 * Called from PipeWire's real-time thread whenever new audio data is needed.
 * We dequeue a buffer, call your audio_callback() to fill it, and then re-queue.
 */
static void on_process(void *userdata) {
	struct pw_buffer *buffer;
	struct spa_buffer *spa_buf;
	int16_t *data;
	uint32_t size;
	uint32_t frames;
	struct pw_time time_info;

	buffer = pw_stream_dequeue_buffer(pa_stream);
	if(!buffer) {
		/* No buffer available, skip. */
		return;
	}

	spa_buf = buffer->buffer;
	if(!spa_buf->datas || !spa_buf->datas[0].data) {
		pw_stream_queue_buffer(pa_stream, buffer);
		return;
	}

	data = spa_buf->datas[0].data;
	size = spa_buf->datas[0].maxsize;
	frames = size / FRAME_SIZE;

	// if(pw_stream_get_time_n(pa_stream, &time_info, sizeof(time_info)) == 0) {
	// 	playback_cursor = time_info.now;
	// }
	// printf("Cursor(ns): %luns\n", playback_cursor);

	audio_callback_thread(data, frames);

	if(spa_buf->datas[0].chunk) {
		spa_buf->datas[0].chunk->size = frames * FRAME_SIZE;
		spa_buf->datas[0].chunk->stride = FRAME_SIZE;
	}

	pw_stream_queue_buffer(pa_stream, buffer);
}

/*
 * Initialize PipeWire, create the stream, and connect for audio playback.
 * Returns immediately so your main thread can continue.
 */
int audio_initialize(void) {
	pw_init(0, 0);

	pa_thread_loop = pw_thread_loop_new("my-audio-loop", 0);
	if(pa_thread_loop) {
		if(pw_thread_loop_start(pa_thread_loop) == 0) {
			pw_thread_loop_lock(pa_thread_loop);

			pa_context = pw_context_new(pw_thread_loop_get_loop(pa_thread_loop), 0, 0);
			if(pa_context) {
				pa_core = pw_context_connect(pa_context, 0, 0);
				if(pa_core){
					static const struct spa_dict_item items[] = {
						SPA_DICT_ITEM_INIT(PW_KEY_MEDIA_TYPE, "Audio"),
						SPA_DICT_ITEM_INIT(PW_KEY_MEDIA_CATEGORY, "Playback"),
						SPA_DICT_ITEM_INIT(PW_KEY_MEDIA_ROLE, "Game"),
						SPA_DICT_ITEM_INIT(PW_KEY_NODE_LATENCY, "512/48000")
					};
					struct pw_properties *props = pw_properties_new_dict(&SPA_DICT_INIT(items, 4));
					// pw_properties_free(props);

					pa_stream = pw_stream_new(pa_core, "My Audio Stream", props);
					if(pa_stream) {
						static struct pw_stream_events stream_events = { PW_VERSION_STREAM_EVENTS, .process = on_process, };
						pw_stream_add_listener(pa_stream, &pa_stream_listener, &stream_events, 0);

						/*
						* Build two SPA params:
						* 1) The audio format: S16_LE, SAMPLE_RATE, NUM_CHANNELS
						* 2) The buffer param: request BUFFER_SIZE bytes per buffer
						*/
						uint8_t fmt_buffer[1024];
						struct spa_pod_builder fmt_builder = SPA_POD_BUILDER_INIT(fmt_buffer, sizeof(fmt_buffer));
						const struct spa_pod *fmt_param = spa_pod_builder_add_object(
							&fmt_builder,
							SPA_TYPE_OBJECT_Format, SPA_PARAM_EnumFormat,
							SPA_FORMAT_mediaType,		SPA_POD_Id(SPA_MEDIA_TYPE_audio),
							SPA_FORMAT_mediaSubtype,	SPA_POD_Id(SPA_MEDIA_SUBTYPE_raw),
							SPA_FORMAT_AUDIO_format,	SPA_POD_Id(SPA_AUDIO_FORMAT_S16_LE),
							SPA_FORMAT_AUDIO_rate,		SPA_POD_Int(SAMPLE_RATE),
							SPA_FORMAT_AUDIO_channels,	SPA_POD_Int(NUM_CHANNELS)
						);

						uint8_t buf_buffer[1024];
						struct spa_pod_builder buf_builder = SPA_POD_BUILDER_INIT(buf_buffer, sizeof(buf_buffer));
						struct spa_pod *buf_param = spa_pod_builder_add_object(
							&buf_builder,
							SPA_TYPE_OBJECT_ParamBuffers, SPA_PARAM_Buffers,
							SPA_PARAM_BUFFERS_buffers,	SPA_POD_CHOICE_RANGE_Int(8, 2, 16),				/* We'll request 8 buffers, each of size = BUFFER_SIZE bytes. */
							SPA_PARAM_BUFFERS_blocks,	SPA_POD_Int(1),
							SPA_PARAM_BUFFERS_size,		SPA_POD_CHOICE_RANGE_Int(BUFFER_SIZE, BUFFER_SIZE, BUFFER_SIZE*8),
							SPA_PARAM_BUFFERS_stride,	SPA_POD_Int(FRAME_SIZE),
							SPA_PARAM_BUFFERS_align,	SPA_POD_Int(16)
						);

						const struct spa_pod *params[2];
						params[0] = fmt_param;
						params[1] = buf_param;

						int res = pw_stream_connect(pa_stream, PW_DIRECTION_OUTPUT, PW_ID_ANY, PW_STREAM_FLAG_AUTOCONNECT | PW_STREAM_FLAG_RT_PROCESS | PW_STREAM_FLAG_MAP_BUFFERS, params, 2);
						pw_thread_loop_unlock(pa_thread_loop);
						return 0;

					} else {
						fprintf(stderr, "Failed to create PipeWire stream\n");
					}
					pw_core_disconnect(pa_core);
				} else {
					fprintf(stderr, "Failed to connect context to core\n");
				}
				pw_context_destroy(pa_context);
			} else {
				fprintf(stderr, "Failed to create PipeWire context\n");
			}
			pw_thread_loop_unlock(pa_thread_loop);
			pw_thread_loop_stop(pa_thread_loop);
		} else {
			fprintf(stderr, "Failed to start PipeWire thread loop\n");
		}
		pw_thread_loop_destroy(pa_thread_loop);
	} else {
		fprintf(stderr, "Failed to create PipeWire thread loop\n");
	}
	pw_deinit();
	return -1;
}

/*
 * Clean up PipeWire objects, stop the thread loop, and deinit.
 * This should be called before your program exits.
 */
void audio_shutdown(void) {
	if(!pa_thread_loop) {
		return;
	}

	pw_thread_loop_lock(pa_thread_loop);

	if(pa_stream){
		pw_stream_disconnect(pa_stream);
		pw_stream_destroy(pa_stream);
	}

	if(pa_core){
		pw_core_disconnect(pa_core);
	}

	if(pa_context){
		pw_context_destroy(pa_context);
	}

	pw_thread_loop_unlock(pa_thread_loop);
	pw_thread_loop_stop(pa_thread_loop);
	pw_thread_loop_destroy(pa_thread_loop);
	pw_deinit();
}















#elif _WIN32

#define COBJMACROS
#include <windows.h>
#include <initguid.h>
#include <audioclient.h>
#include <mmdeviceapi.h>
#include <avrt.h>
#include <stdint.h>
#include <stdio.h>
#include <timeapi.h>

/*
 * Minimal WASAPI shared-mode audio playback with explicit 48kHz/16-bit/2ch.
 */

#define NUM_CHANNELS 2

static IMMDeviceEnumerator *enumerator;
static IMMDevice *device_out;
static IAudioClient *audio_client_out;
static IAudioRenderClient *render_client;
static HANDLE audio_event;
static HANDLE audio_thread;
static int running;

static DWORD WINAPI audio_thread_proc(void *arg) {
	UINT32 buffer_size;
	UINT32 padding;
	UINT32 available;
	uint8_t *data;

	IAudioClient_GetBufferSize(audio_client_out, &buffer_size);

	while(running) {
		WaitForSingleObject(audio_event, INFINITE);
		if(!running) {
			break;
		}

		IAudioClient_GetCurrentPadding(audio_client_out, &padding);
		available = buffer_size - padding;
		IAudioRenderClient_GetBuffer(render_client, available, &data);
		audio_callback_thread((int16_t*)data, available);
		IAudioRenderClient_ReleaseBuffer(render_client, available, 0);
	}
	return 0;
}

void audio_initialize() {
	WAVEFORMATEX wf;
	REFERENCE_TIME dur_out;

	CoInitializeEx(0, COINIT_MULTITHREADED);
	if(SUCCEEDED(CoCreateInstance(&CLSID_MMDeviceEnumerator, 0, CLSCTX_ALL, &IID_IMMDeviceEnumerator, (void**)&enumerator))) {
		if(SUCCEEDED(IMMDeviceEnumerator_GetDefaultAudioEndpoint(enumerator, eRender, eConsole, &device_out))) {
			if(SUCCEEDED(IMMDevice_Activate(device_out, &IID_IAudioClient, CLSCTX_ALL, 0, (void**)&audio_client_out))) {
				wf.wFormatTag = WAVE_FORMAT_PCM;
				wf.nChannels = NUM_CHANNELS;
				wf.nSamplesPerSec = 48000;
				wf.wBitsPerSample = 16;
				wf.nBlockAlign = (wf.nChannels * wf.wBitsPerSample) / 8;
				wf.nAvgBytesPerSec = wf.nSamplesPerSec * wf.nBlockAlign;
				wf.cbSize = 0;

				IAudioClient_GetDevicePeriod(audio_client_out, &dur_out, 0);
				IAudioClient_Initialize(audio_client_out, AUDCLNT_SHAREMODE_SHARED, AUDCLNT_STREAMFLAGS_EVENTCALLBACK, dur_out, 0, &wf, 0);
				audio_event = CreateEvent(0, FALSE, FALSE, 0);
				if(audio_event){
					IAudioClient_SetEventHandle(audio_client_out, audio_event);
					IAudioClient_GetService(audio_client_out, &IID_IAudioRenderClient, (void**)&render_client);
					IAudioClient_Start(audio_client_out);

					running = 1;
					audio_thread = CreateThread(0, 0, audio_thread_proc, 0, 0, 0);
					return;
				} else {
					printf("Failed to create audio event\n");
				}
				audio_client_out->lpVtbl->Release(audio_client_out);
			} else {
				printf("Failed to activate audio client\n");
			}
			device_out->lpVtbl->Release(device_out);
		} else {
			printf("Failed to get default audio endpoint\n");
		}
		enumerator->lpVtbl->Release(enumerator);
	} else {
		printf("Failed to create MMDeviceEnumerator\n");
	}
}

void audio_shutdown() {
	running = 0;
	if(audio_thread) {
		SetEvent(audio_event);
		WaitForSingleObject(audio_thread, INFINITE);
		CloseHandle(audio_thread);
	}
	if(audio_event) {
		CloseHandle(audio_event);
	}
	if(audio_client_out) {
		IAudioClient_Stop(audio_client_out);
		audio_client_out->lpVtbl->Release(audio_client_out);
	}
	if(render_client) {
		render_client->lpVtbl->Release(render_client);
	}
	if(device_out) {
		device_out->lpVtbl->Release(device_out);
	}
	if(enumerator) {
		enumerator->lpVtbl->Release(enumerator);
	}
	CoUninitialize();
}

#endif





// BELOW IS FOR FUTURE FRAME SYNCHRONIZATION!!!

#if 0
// Audio sync throttling logic (using audio playback clock)

#define AUDIO_SAMPLE_RATE 48000
#define FRAMETIME (1000000000 / 60) // NES: ~16.67ms per frame (replace as needed for PAL/other)

static uint64_t emulator_start_time_ns = 0;
static uint64_t audio_start_time_ns = 0;

// Stub: return current audio playback time in nanoseconds
uint64_t get_audio_playback_time_ns(void);

// Call this once at emulation start
void audio_sync_init(uint64_t current_time_ns) {
	emulator_start_time_ns = current_time_ns;
	audio_start_time_ns = get_audio_playback_time_ns();
}

// Call this at the end of each frame
void audio_throttle_emulator(uint64_t frame_number, int64_t *frame_duration_ns) {
	uint64_t expected_emulated_time = frame_number * FRAMETIME;
	uint64_t actual_audio_time = get_audio_playback_time_ns() - audio_start_time_ns;

	int64_t drift = (int64_t)(actual_audio_time) - (int64_t)(expected_emulated_time);

	// Adjust frame duration to correct drift gradually
	*frame_duration_ns -= drift / 8;
	// Clamp adjustment to avoid jitter
	if(*frame_duration_ns > FRAMETIME + 50000) {
		*frame_duration_ns = FRAMETIME + 50000;
	} else if(*frame_duration_ns < FRAMETIME - 50000) {
		*frame_duration_ns = FRAMETIME - 50000;
	}
}

#ifdef _WIN32
#include <windows.h>
#include <mmdeviceapi.h>
#include <audioclient.h>

uint64_t get_audio_playback_time_ns(void) {
	// WASAPI: query IAudioClock interface
	// This is just a placeholder. You’ll need to cache IAudioClock *audio_clock externally.
	extern IAudioClock *audio_clock;
	UINT64 pos;
	audio_clock->lpVtbl->GetPosition(audio_clock, &pos, 0);
	return (pos * 1000000000ULL) / AUDIO_SAMPLE_RATE;
}

#else
// PipeWire backend
#include <spa/clock/clock.h>
extern struct spa_clock *audio_clock;

uint64_t get_audio_playback_time_ns(void) {
	struct spa_clock_info info;
	audio_clock->get_time(audio_clock, &info);
	return info.nsec;
}
#endif

#endif