looper/backends/playback/zsm/zsm_backend.hpp

#pragma once
#include "playback_backend.hpp"
#include <omp.h>
#include "x16emu/ymglue.h"
#include <cstdint>
#include <stddef.h>
#include <stdint.h>
#include <stdlib.h>
#include <string.h>
#include <streambuf>
#include <vector>
#include <util.hpp>
#include <SDL.h>
extern "C" {
#include "x16emu/audio.h"
#include "x16emu/vera_pcm.h"
#include "x16emu/vera_psg.h"
#include "x16emu/ymglue.h"
}
#include "file_backend.hpp"
#define YM_FREQ (3579545/64)
#define PSG_FREQ (AUDIO_SAMPLERATE)
enum ZsmCommandId {
    PsgWrite,
    ExtCmd,
    FmWrite,
    ZsmEOF,
    Delay
};
struct reg_pair {
    uint8_t reg;
    uint8_t val;
};
struct ZsmCommand {
    ZsmCommandId id;
    union {
        struct {
            uint8_t reg;
            uint8_t val;
        } psg_write;
        struct {
            uint8_t channel;
            uint8_t bytes;
            union {
                uint8_t *pcm;
                struct {
                    uint8_t chip_id;
                    uint8_t writes;
                    uint8_t *write_bytes;
                } expansion;
                uint8_t *sync;
                uint8_t *custom;
            };
        } extcmd;
        struct {
            uint8_t len;
            reg_pair *regs;
        } fm_write;
        uint8_t delay;
    };
    ~ZsmCommand();
};
struct pcm_instrument {
	uint8_t geom;
	uint32_t loop_rem;
	uint32_t loop;
	bool islooped;
	uint32_t remain;
	uint8_t *data = nullptr;
	inline ~pcm_instrument() {
		if (data != nullptr) free((void*)data);
	}
};
class ZsmBackend : public PlaybackBackend {
    File *file;
    Fifo<int16_t> audio_buf;
    DynPtr psg_buf;
    DynPtr pcm_buf;
    DynPtr out_buf;
    DynPtr ym_buf;
    DynPtr ym_resample_buf;
    bool ym_recorded = false;
    uint8_t ym_data[256];
    uint32_t loop_rem;
    uint32_t pcm_data_offs;
    uint8_t pcm_data_instruments;
    uint32_t loop;
    bool islooped;
    uint32_t remain;
    uint32_t cur;
    uint32_t pcm_loop_point;
    uint32_t rem_point;
    SDL_AudioStream *fm_stream;
    std::vector<pcm_instrument*> instruments;
    uint8_t *audio_sample = nullptr;
    int16_t combine_audio(int16_t a, int16_t b) {
        return (int16_t)((((int32_t)a) + ((int32_t)b)) >> 1);
    }
    void audio_step(size_t samples) {
    	if (samples == 0) return;
     	while (remain != 0 && pcm_fifo_avail() < samples) {
     		if (pcm_read_rate() == 0) break;
            if ((--remain) == 0) {
                if (islooped) {
                 	cur = loop;
                    remain = loop_rem;
                } else {
                	break;
                }
            }
            size_t oldpos = file->get_pos();
            uint8_t sample = audio_sample[cur++];
            pcm_write_fifo(sample);
        }
        samples *= 2;
    	int16_t *psg_ptr = psg_buf.get_item_sized<int16_t>(samples);
        int16_t *pcm_ptr = pcm_buf.get_item_sized<int16_t>(samples);
    	psg_render(psg_ptr, samples / 2);
        pcm_render(pcm_ptr, samples / 2);
        int16_t *out_ptr = out_buf.get_item_sized<int16_t>(samples);
        // The exact amount of samples needed for the stream.
        double ratio = ((double)YM_FREQ) / ((double)PSG_FREQ);
        size_t needed_samples = ((size_t)std::floor(samples * ratio)) / 2;
        int16_t *ym_ptr = ym_buf.get_item_sized<int16_t>(needed_samples * 2);
        YM_stream_update(ym_ptr, needed_samples);
        assert(SDL_AudioStreamPut(fm_stream, ym_ptr, needed_samples * 2 * sizeof(int16_t)) == 0);
        while (SDL_AudioStreamAvailable(fm_stream) < ((samples + 2) * sizeof(int16_t))) {
            YM_stream_update(ym_ptr, 1);
            assert(SDL_AudioStreamPut(fm_stream, ym_ptr, 2 * sizeof(int16_t)) == 0);
        }
        int16_t *ym_resample_ptr = ym_resample_buf.get_item_sized<int16_t>(samples);
        ssize_t ym_resample_len = SDL_AudioStreamGet(fm_stream, ym_resample_ptr, (samples + 2) * sizeof(int16_t));
        assert(ym_resample_len >= 0);
        ym_resample_len /= sizeof(int16_t);
        for (size_t i = 0; i < samples / 2; i++) {
            size_t j = i * 2;
            int32_t psg[2] = {psg_ptr[j], psg_ptr[j + 1]};
            int32_t pcm[2] = {pcm_ptr[j], pcm_ptr[j + 1]};
            int32_t vera[2] = {psg[0] + pcm[0], psg[1] + pcm[1]};
            int32_t fm[2] = {ym_resample_ptr[j], ym_resample_ptr[j + 1]};
            int16_t mix[2] = {(vera[0] + fm[0]) / 3, (vera[1] + fm[1]) / 3};
            out_ptr[j++] = mix[0];
            out_ptr[j++] = mix[1];
        }
        audio_buf.push(out_ptr, samples);
    }
    inline void *reserve(size_t len) {
       	return (void*)audio_buf.reserve(len);
    }
    inline size_t copy_out(void *buf, size_t len) {
    	return audio_buf.pop((int16_t*)buf, len);
    }
    uint32_t loop_point;
    double loop_pos = 0.0;
    uint32_t pcm_offset;
    uint8_t fm_mask;
    uint16_t psg_channel_mask;
    uint16_t tick_rate;
    size_t music_data_start;
    size_t music_data_len;
    double ticks;
    ssize_t delayTicks = 0;
    double position;
    double cpuClocks = 0;
    inline double get_delay_per_frame() {
        return 1.0;
    }
    void tick(bool step = true);\
    void seek_internal(double position, bool loop = true);
    ZsmCommand get_command();
    public:
    uint64_t get_min_samples() override;
    std::optional<uint64_t> get_max_samples() override;
    inline std::string get_id() override {
        return "zsm";
    }
    inline std::string get_name() override {
        return "ZSM player";
    }
    void seek(double position) override;
    void load(const char *filename) override;
    void switch_stream(int idx) override;
    void cleanup() override;
    int get_stream_idx() override;
    size_t render(void *buf, size_t maxlen) override;
    double get_position() override;
    inline double get_length() override {
        return length;
    }
    inline ~ZsmBackend() override { }
};