diff --git a/src/azurespeechapi.cpp b/src/azurespeechapi.cpp
index a5c61cf..36d900a 100644
--- a/src/azurespeechapi.cpp
+++ b/src/azurespeechapi.cpp
@@ -1,5 +1,4 @@
 #include "azurespeechapi.h"
-#include <QDebug>
 
 using namespace Microsoft::CognitiveServices::Speech;
 using namespace Microsoft::CognitiveServices::Speech::Translation;
diff --git a/src/main.cpp b/src/main.cpp
index 60ca3ae..cf05e90 100644
--- a/src/main.cpp
+++ b/src/main.cpp
@@ -4,7 +4,6 @@
 #include <QDateTime>
 #include <windows.h>
 #include <dbghelp.h>
-#include <QDebug>
 #include "mainwindow.h"
 #include "logger.h"
 
diff --git a/src/wasapiaudiocapture.cpp b/src/wasapiaudiocapture.cpp
index dd7a58e..b77b8fb 100644
--- a/src/wasapiaudiocapture.cpp
+++ b/src/wasapiaudiocapture.cpp
@@ -1,5 +1,8 @@
 #include "wasapiaudiocapture.h"
-#include <QDebug>
+#include "logger.h"
+#include <comdef.h>
+#include <vector>
+#include <cmath>
 
 WasapiAudioCapture::WasapiAudioCapture(QObject *parent)
     : QObject(parent)
@@ -11,23 +14,31 @@ WasapiAudioCapture::WasapiAudioCapture(QObject *parent)
     , m_isCapturing(false)
     , m_waveFormat(nullptr)
     , m_bufferFrameCount(0)
+    , logger(std::make_unique<Logger>())
 {
+    LOG_INFO("WasapiAudioCapture 初始化");
 }
 
 WasapiAudioCapture::~WasapiAudioCapture()
 {
+    LOG_INFO("WasapiAudioCapture 析构");
     stopCapture();
     cleanupWASAPI();
 }
 
 bool WasapiAudioCapture::initializeWASAPI()
 {
+    LOG_INFO("开始初始化 WASAPI");
     HRESULT hr = CoCreateInstance(__uuidof(MMDeviceEnumerator),
                                  nullptr,
                                  CLSCTX_ALL,
                                  __uuidof(IMMDeviceEnumerator),
                                  (void**)&m_deviceEnumerator);
     if (FAILED(hr)) {
+        _com_error err(hr);
+        LOG_ERROR(QString("无法创建设备枚举器，错误代码: 0x%1, 描述: %2")
+                 .arg(hr, 8, 16, QChar('0'))
+                 .arg(QString::fromWCharArray(err.Description())));
         emit error("无法创建设备枚举器");
         return false;
     }
@@ -35,6 +46,10 @@ bool WasapiAudioCapture::initializeWASAPI()
     // 获取默认音频输出设备
     hr = m_deviceEnumerator->GetDefaultAudioEndpoint(eRender, eConsole, &m_audioDevice);
     if (FAILED(hr)) {
+        _com_error err(hr);
+        LOG_ERROR(QString("无法获取默认音频输出设备，错误代码: 0x%1, 描述: %2")
+                 .arg(hr, 8, 16, QChar('0'))
+                 .arg(QString::fromWCharArray(err.Description())));
         emit error("无法获取默认音频输出设备");
         return false;
     }
@@ -45,28 +60,81 @@ bool WasapiAudioCapture::initializeWASAPI()
                                 nullptr,
                                 (void**)&m_audioClient);
     if (FAILED(hr)) {
+        _com_error err(hr);
+        LOG_ERROR(QString("无法激活音频客户端，错误代码: 0x%1, 描述: %2")
+                 .arg(hr, 8, 16, QChar('0'))
+                 .arg(QString::fromWCharArray(err.Description())));
         emit error("无法激活音频客户端");
         return false;
     }
 
-    // 设置音频格式
-    m_waveFormat = new WAVEFORMATEX();
-    m_waveFormat->wFormatTag = WAVE_FORMAT_PCM;
-    m_waveFormat->nChannels = CHANNELS;
-    m_waveFormat->nSamplesPerSec = SAMPLE_RATE;
-    m_waveFormat->wBitsPerSample = BITS_PER_SAMPLE;
-    m_waveFormat->nBlockAlign = (m_waveFormat->nChannels * m_waveFormat->wBitsPerSample) / 8;
-    m_waveFormat->nAvgBytesPerSec = m_waveFormat->nSamplesPerSec * m_waveFormat->nBlockAlign;
-    m_waveFormat->cbSize = 0;
+    // 获取系统混音格式（可能是 WAVEFORMATEXTENSIBLE）
+    WAVEFORMATEX* pMixFormat = nullptr;
+    hr = m_audioClient->GetMixFormat(&pMixFormat);
+    if (FAILED(hr)) {
+        _com_error err(hr);
+        LOG_ERROR(QString("无法获取设备混音格式，错误代码: 0x%1, 描述: %2")
+                 .arg(hr, 8, 16, QChar('0'))
+                 .arg(QString::fromWCharArray(err.Description())));
+        emit error("无法获取设备混音格式");
+        return false;
+    }
+
+    // 检查是否支持 16kHz/16bit/单声道格式
+    WAVEFORMATEX desiredFormat = {0};
+    desiredFormat.wFormatTag = WAVE_FORMAT_PCM;
+    desiredFormat.nChannels = 1;
+    desiredFormat.nSamplesPerSec = 16000;
+    desiredFormat.wBitsPerSample = 16;
+    desiredFormat.nBlockAlign = desiredFormat.nChannels * desiredFormat.wBitsPerSample / 8;
+    desiredFormat.nAvgBytesPerSec = desiredFormat.nSamplesPerSec * desiredFormat.nBlockAlign;
+
+    BOOL supported = FALSE;
+    hr = m_audioClient->IsFormatSupported(AUDCLNT_SHAREMODE_SHARED, &desiredFormat, nullptr);
+    if (SUCCEEDED(hr)) {
+        LOG_INFO("设备支持 16kHz/16bit/单声道格式，将使用此格式");
+        // 分配足够的内存来存储 WAVEFORMATEXTENSIBLE
+        m_waveFormat = (WAVEFORMATEXTENSIBLE*)CoTaskMemAlloc(sizeof(WAVEFORMATEXTENSIBLE));
+        if (!m_waveFormat) {
+            LOG_ERROR("内存分配失败");
+            CoTaskMemFree(pMixFormat);
+            emit error("内存分配失败");
+            return false;
+        }
+        // 复制基本格式信息
+        memcpy(&m_waveFormat->Format, &desiredFormat, sizeof(WAVEFORMATEX));
+        // 设置扩展信息
+        m_waveFormat->Format.cbSize = sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX);
+        m_waveFormat->Samples.wValidBitsPerSample = desiredFormat.wBitsPerSample;
+        m_waveFormat->dwChannelMask = SPEAKER_FRONT_CENTER;
+        m_waveFormat->SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
+    } else {
+        LOG_INFO(QString("设备不支持 16kHz/16bit/单声道格式，将使用系统混音格式：%1 通道, %2 Hz, %3 bit")
+                 .arg(pMixFormat->nChannels)
+                 .arg(pMixFormat->nSamplesPerSec)
+                 .arg(pMixFormat->wBitsPerSample));
+        // 直接使用系统混音格式
+        m_waveFormat = (WAVEFORMATEXTENSIBLE*)pMixFormat;
+        pMixFormat = nullptr;  // 防止在后面的 CoTaskMemFree 中重复释放
+    }
 
-    // 初始化音频客户端
+    // 使用选定的格式初始化
     hr = m_audioClient->Initialize(AUDCLNT_SHAREMODE_SHARED,
                                  AUDCLNT_STREAMFLAGS_LOOPBACK,
-                                 0,
-                                 0,
-                                 m_waveFormat,
+                                 0,    // 默认缓冲区时长
+                                 0,    // event-driven 模式
+                                 (WAVEFORMATEX*)m_waveFormat,
                                  nullptr);
+
+    if (pMixFormat) {
+        CoTaskMemFree(pMixFormat);  // 如果之前没有使用系统混音格式，释放它
+    }
+
     if (FAILED(hr)) {
+        _com_error err(hr);
+        LOG_ERROR(QString("无法初始化音频客户端，错误代码: 0x%1, 描述: %2")
+                 .arg(hr, 8, 16, QChar('0'))
+                 .arg(QString::fromWCharArray(err.Description())));
         emit error("无法初始化音频客户端");
         return false;
     }
@@ -74,23 +142,35 @@ bool WasapiAudioCapture::initializeWASAPI()
     // 获取缓冲区大小
     hr = m_audioClient->GetBufferSize(&m_bufferFrameCount);
     if (FAILED(hr)) {
+        _com_error err(hr);
+        LOG_ERROR(QString("无法获取缓冲区大小，错误代码: 0x%1, 描述: %2")
+                 .arg(hr, 8, 16, QChar('0'))
+                 .arg(QString::fromWCharArray(err.Description())));
         emit error("无法获取缓冲区大小");
         return false;
     }
 
+    LOG_INFO(QString("音频缓冲区大小：%1 帧").arg(m_bufferFrameCount));
+
     // 获取捕获客户端
     hr = m_audioClient->GetService(__uuidof(IAudioCaptureClient),
                                   (void**)&m_captureClient);
     if (FAILED(hr)) {
+        _com_error err(hr);
+        LOG_ERROR(QString("无法获取捕获客户端，错误代码: 0x%1, 描述: %2")
+                 .arg(hr, 8, 16, QChar('0'))
+                 .arg(QString::fromWCharArray(err.Description())));
         emit error("无法获取捕获客户端");
         return false;
     }
 
+    LOG_INFO("WASAPI 初始化完成");
     return true;
 }
 
 void WasapiAudioCapture::cleanupWASAPI()
 {
+    LOG_INFO("开始清理 WASAPI 资源");
     if (m_captureClient) {
         m_captureClient->Release();
         m_captureClient = nullptr;
@@ -108,23 +188,28 @@ void WasapiAudioCapture::cleanupWASAPI()
         m_deviceEnumerator = nullptr;
     }
     if (m_waveFormat) {
-        delete m_waveFormat;
+        CoTaskMemFree(m_waveFormat);  // 使用 CoTaskMemFree 释放内存
         m_waveFormat = nullptr;
     }
+    LOG_INFO("WASAPI 资源清理完成");
 }
 
 bool WasapiAudioCapture::startCapture()
 {
     if (m_isCapturing) {
+        LOG_INFO("音频捕获已经在进行中");
         return true;
     }
 
+    LOG_INFO("开始音频捕获");
     if (!initializeWASAPI()) {
+        LOG_ERROR("初始化 WASAPI 失败");
         return false;
     }
 
     HRESULT hr = m_audioClient->Start();
     if (FAILED(hr)) {
+        LOG_ERROR("无法启动音频捕获");
         emit error("无法启动音频捕获");
         return false;
     }
@@ -132,11 +217,13 @@ bool WasapiAudioCapture::startCapture()
     m_isCapturing = true;
     m_captureThread = CreateThread(nullptr, 0, captureThread, this, 0, nullptr);
     if (!m_captureThread) {
+        LOG_ERROR("无法创建捕获线程");
         emit error("无法创建捕获线程");
         stopCapture();
         return false;
     }
 
+    LOG_INFO("音频捕获已启动");
     return true;
 }
 
@@ -146,6 +233,7 @@ void WasapiAudioCapture::stopCapture()
         return;
     }
 
+    LOG_INFO("停止音频捕获");
     m_isCapturing = false;
     if (m_captureThread) {
         WaitForSingleObject(m_captureThread, INFINITE);
@@ -156,6 +244,7 @@ void WasapiAudioCapture::stopCapture()
     if (m_audioClient) {
         m_audioClient->Stop();
     }
+    LOG_INFO("音频捕获已停止");
 }
 
 DWORD WINAPI WasapiAudioCapture::captureThread(LPVOID context)
@@ -205,7 +294,83 @@ void WasapiAudioCapture::processAudioData(const BYTE* data, UINT32 numFrames)
         return;
     }
 
-    UINT32 dataSize = numFrames * m_waveFormat->nBlockAlign;
-    QByteArray audioData(reinterpret_cast<const char*>(data), dataSize);
-    emit audioDataReceived(audioData);
+    // 将 BYTE* 转换为 float* 数组
+    auto floatBuf = reinterpret_cast<const float*>(data);
+    
+    // 创建 16-bit PCM 缓冲区
+    std::vector<int16_t> pcm16;
+    pcm16.reserve(numFrames * m_waveFormat->Format.nChannels);
+
+    // 转换每个采样点
+    for (UINT32 i = 0; i < numFrames * m_waveFormat->Format.nChannels; ++i) {
+        float f = floatBuf[i];
+        // 裁剪到 [-1.0, 1.0] 范围
+        if (f > 1.f)  f = 1.f;
+        if (f < -1.f) f = -1.f;
+        // 转换为 16-bit PCM
+        pcm16.push_back(static_cast<int16_t>(f * 32767));
+    }
+
+    // 如果当前不是 16kHz/16bit/单声道，需要进行转换
+    std::vector<int16_t> finalBuffer;
+    if (m_waveFormat->Format.nSamplesPerSec != 16000 || m_waveFormat->Format.nChannels != 1) {
+        // 先进行声道混音（如果需要）
+        std::vector<int16_t> monoBuffer;
+        if (m_waveFormat->Format.nChannels > 1) {
+            monoBuffer.reserve(numFrames);
+            for (UINT32 i = 0; i < numFrames; ++i) {
+                int32_t sum = 0;
+                for (UINT32 ch = 0; ch < m_waveFormat->Format.nChannels; ++ch) {
+                    sum += pcm16[i * m_waveFormat->Format.nChannels + ch];
+                }
+                monoBuffer.push_back(static_cast<int16_t>(sum / m_waveFormat->Format.nChannels));
+            }
+        } else {
+            monoBuffer = std::move(pcm16);
+        }
+
+        // 然后进行重采样（如果需要）
+        if (m_waveFormat->Format.nSamplesPerSec != 16000) {
+            const double ratio = 16000.0 / m_waveFormat->Format.nSamplesPerSec;
+            const size_t newSize = static_cast<size_t>(std::ceil(monoBuffer.size() * ratio));
+            finalBuffer.reserve(newSize);
+
+            // 线性插值重采样
+            for (size_t i = 0; i < newSize; ++i) {
+                double pos = i / ratio;
+                size_t pos1 = static_cast<size_t>(std::floor(pos));
+                size_t pos2 = pos1 + 1;
+                double frac = pos - pos1;
+
+                // 边界检查
+                if (pos2 >= monoBuffer.size()) {
+                    pos2 = monoBuffer.size() - 1;
+                }
+
+                // 线性插值
+                double sample = monoBuffer[pos1] * (1.0 - frac) + monoBuffer[pos2] * frac;
+                finalBuffer.push_back(static_cast<int16_t>(std::round(sample)));
+            }
+        } else {
+            finalBuffer = std::move(monoBuffer);
+        }
+    } else {
+        finalBuffer = std::move(pcm16);
+    }
+
+    // 创建输出数据
+    QByteArray out(reinterpret_cast<const char*>(finalBuffer.data()), finalBuffer.size() * sizeof(int16_t));
+    
+    // 使用静态计数器来减少日志输出频率
+    static int logCounter = 0;
+    if (++logCounter % 100 == 0) {  // 每100个数据块记录一次
+        LOG_INFO(QString("处理音频数据：输入 %1 帧，%2 通道，%3 Hz，输出 %4 帧，16kHz/16bit/单声道，输出大小：%5 字节")
+                .arg(numFrames)
+                .arg(m_waveFormat->Format.nChannels)
+                .arg(m_waveFormat->Format.nSamplesPerSec)
+                .arg(finalBuffer.size())
+                .arg(out.size()));
+    }
+
+    emit audioDataReceived(out);
 } 
\ No newline at end of file
diff --git a/src/wasapiaudiocapture.h b/src/wasapiaudiocapture.h
index 82242b7..b405d97 100644
--- a/src/wasapiaudiocapture.h
+++ b/src/wasapiaudiocapture.h
@@ -7,6 +7,8 @@
 #include <mmdeviceapi.h>
 #include <audioclient.h>
 #include <functiondiscoverykeys_devpkey.h>
+#include "logger.h"
+#include <memory>
 
 class WasapiAudioCapture : public QObject
 {
@@ -35,8 +37,9 @@ class WasapiAudioCapture : public QObject
     IAudioCaptureClient* m_captureClient;
     HANDLE m_captureThread;
     bool m_isCapturing;
-    WAVEFORMATEX* m_waveFormat;
+    WAVEFORMATEXTENSIBLE* m_waveFormat;
     UINT32 m_bufferFrameCount;
+    std::unique_ptr<Logger> logger;
     static const int SAMPLE_RATE = 16000;
     static const int CHANNELS = 1;
     static const int BITS_PER_SAMPLE = 16;