git clone https://gitcode.com/gh_mirrors/em/emscripten
cd emscripten
./bootstrap
source ./emsdk_env.sh

# 運行SIMD測試驗證環境
make test SIMD=1

void noise_reduction(float* input, float* output, float* noise_profile, int length) {
  for (int i = 0; i < length; i++) {
    // 計算幅度譜
    float mag = sqrt(input[2*i]*input[2*i] + input[2*i+1]*input[2*i+1]);
    // 頻譜減法
    float mag_clean = max(mag - noise_profile[i], 0.0f);
    // 相位保留
    output[2*i] = mag_clean * cos(input[2*i+1]);
    output[2*i+1] = mag_clean * sin(input[2*i+1]);
  }
}

#include <emmintrin.h>  // SSE2指令集

void noise_reduction_simd(float* input, float* output, float* noise_profile, int length) {
  __m128 zero = _mm_setzero_ps();
  
  for (int i = 0; i < length; i += 4) {
    // 加載4個複數樣本 (8個float)
    __m128 in1 = _mm_load_ps(&input[2*i]);
    __m128 in2 = _mm_load_ps(&input[2*i+4]);
    
    // 計算幅度: sqrt(re^2 + im^2)
    __m128 re = _mm_shuffle_ps(in1, in2, _MM_SHUFFLE(0, 2, 0, 2));
    __m128 im = _mm_shuffle_ps(in1, in2, _MM_SHUFFLE(1, 3, 1, 3));
    __m128 re_sq = _mm_mul_ps(re, re);
    __m128 im_sq = _mm_mul_ps(im, im);
    __m128 mag = _mm_sqrt_ps(_mm_add_ps(re_sq, im_sq));
    
    // 頻譜減法: max(mag - noise_profile, 0)
    __m128 noise = _mm_load_ps(&noise_profile[i]);
    __m128 mag_clean = _mm_max_ps(_mm_sub_ps(mag, noise), zero);
    
    // 相位保留並存儲結果
    // ...實現省略...
  }
}

emcc audio_processor.cpp -o audio_processor.js \
  -O3 \
  -msimd128 \                # 啓用WebAssembly SIMD
  -mssse3 \                  # 啓用SSSE3指令集
  -ffast-math \              # 允許激進的數學優化
  -s ALLOW_MEMORY_GROWTH=1 \ # 允許內存增長
  -s EXPORTED_FUNCTIONS="['_init_audio','_process_audio']" \
  -s EXTRA_EXPORTED_RUNTIME_METHODS="['ccall','cwrap']"

emcc -S -msimd128 -O3 audio_processor.cpp -o audio_processor.s

#include <emscripten.h>
#include <emmintrin.h>

// FIR濾波器係數
const float fir_coefficients[] = {0.1, 0.2, 0.3, 0.2, 0.1};
const int filter_taps = 5;

// 音頻處理緩衝區
float input_buffer[1024];
float output_buffer[1024];
float delay_line[filter_taps + 1023];
int delay_index = 0;

// SIMD優化的FIR濾波函數
void fir_filter_simd(float* input, float* output, int length) {
  __m128 coeffs[filter_taps/4 + 1];
  
  // 加載濾波器係數到SIMD寄存器
  for (int i = 0; i < filter_taps; i += 4) {
    coeffs[i/4] = _mm_load_ps(&fir_coefficients[i]);
  }
  
  // 處理每個輸入樣本
  for (int i = 0; i < length; i++) {
    // 更新延遲線
    delay_line[delay_index] = input[i];
    delay_index = (delay_index + 1) % (filter_taps + length);
    
    __m128 sum = _mm_setzero_ps();
    
    // SIMD點積運算
    for (int t = 0; t < filter_taps; t += 4) {
      int idx = (delay_index - t + filter_taps + length) % (filter_taps + length);
      __m128 samples = _mm_load_ps(&delay_line[idx]);
      __m128 products = _mm_mul_ps(samples, coeffs[t/4]);
      sum = _mm_add_ps(sum, products);
    }
    
    // 水平求和並存儲結果
    float result;
    _mm_store_ss(&result, sum);
    output[i] = result;
  }
}

// 暴露給JavaScript的處理函數
EMSCRIPTEN_KEEPALIVE
void process_audio(int input_ptr, int output_ptr, int length) {
  float* input = reinterpret_cast<float*>(input_ptr);
  float* output = reinterpret_cast<float*>(output_ptr);
  
  // 使用SIMD優化的FIR濾波器處理音頻
  fir_filter_simd(input, output, length);
  
  // 其他音頻效果處理...
}

EMSCRIPTEN_KEEPALIVE
int init_audio(int sample_rate, int channels) {
  // 初始化音頻處理參數
  return 0;
}

// 初始化音頻上下文
const audioContext = new AudioContext({ sampleRate: 44100 });

// 通過Emscripten cwrap包裝C函數
const initAudio = Module.cwrap('init_audio', 'number', ['number', 'number']);
const processAudio = Module.cwrap('process_audio', 'void', ['number', 'number', 'number']);

// 初始化音頻處理器
initAudio(44100, 2);

// 創建ScriptProcessorNode
const scriptProcessor = audioContext.createScriptProcessor(1024, 2, 2);

// 音頻處理回調
scriptProcessor.onaudioprocess = function(e) {
  const inputL = e.inputBuffer.getChannelData(0);
  const inputR = e.inputBuffer.getChannelData(1);
  const outputL = e.outputBuffer.getChannelData(0);
  const outputR = e.outputBuffer.getChannelData(1);
  
  // 將輸入數據複製到WebAssembly堆
  const inputPtr = Module._malloc(inputL.length * 4);
  const outputPtr = Module._malloc(outputL.length * 4);
  
  Module.HEAPF32.set(inputL, inputPtr / 4);
  
  // 調用C處理函數
  processAudio(inputPtr, outputPtr, inputL.length);
  
  // 將結果複製回輸出緩衝區
  outputL.set(Module.HEAPF32.subarray(outputPtr / 4, outputPtr / 4 + inputL.length));
  
  // 釋放內存
  Module._free(inputPtr);
  Module._free(outputPtr);
};

// 連接音頻節點
navigator.mediaDevices.getUserMedia({ audio: true })
  .then(stream => {
    const source = audioContext.createMediaStreamSource(stream);
    source.connect(scriptProcessor);
    scriptProcessor.connect(audioContext.destination);
  });