reverted rewrite of ppu, optimized what functions should be forced inline, gained ~2.5% performance

1 files changed, 0 insertions, 458 deletions

diff --git a/base/audio.c b/base/audio.c
deleted file mode 100644
index 1d385d5..0000000
--- a/base/audio.c
+++ /dev/null
@@ -1,458 +0,0 @@
-#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