diff --git a/main/CMakeLists.txt b/main/CMakeLists.txt
index 34bdb8d7..0318e4ea 100644
--- a/main/CMakeLists.txt
+++ b/main/CMakeLists.txt
@@ -4,6 +4,7 @@ set(SOURCES "audio_codecs/audio_codec.cc"
             "audio_codecs/es8311_audio_codec.cc"
             "audio_codecs/es8374_audio_codec.cc"
             "audio_codecs/es8388_audio_codec.cc"
+            "audio_processing/audio_debugger.cc"
             "led/single_led.cc"
             "led/circular_strip.cc"
             "led/gpio_led.cc"
diff --git a/main/Kconfig.projbuild b/main/Kconfig.projbuild
index 739f5b22..f85c9e1f 100644
--- a/main/Kconfig.projbuild
+++ b/main/Kconfig.projbuild
@@ -392,6 +392,19 @@ config USE_SERVER_AEC
     help
         启用服务器端 AEC，需要服务器支持
 
+config USE_AUDIO_DEBUGGER
+    bool "Enable Audio Debugger"
+    default n
+    help
+        启用音频调试功能，通过UDP发送音频数据
+
+config AUDIO_DEBUG_UDP_SERVER
+    string "Audio Debug UDP Server Address"
+    default "192.168.2.100:8000"
+    depends on USE_AUDIO_DEBUGGER
+    help
+        UDP服务器地址，格式: IP:PORT，用于接收音频调试数据
+
 choice IOT_PROTOCOL
     prompt "IoT Protocol"
     default IOT_PROTOCOL_MCP
diff --git a/main/application.cc b/main/application.cc
index 53aa8c3e..42c05469 100644
--- a/main/application.cc
+++ b/main/application.cc
@@ -10,6 +10,7 @@
 #include "iot/thing_manager.h"
 #include "assets/lang_config.h"
 #include "mcp_server.h"
+#include "audio_debugger.h"
 
 #if CONFIG_USE_AUDIO_PROCESSOR
 #include "afe_audio_processor.h"
@@ -569,6 +570,7 @@ void Application::Start() {
     });
     bool protocol_started = protocol_->Start();
 
+    audio_debugger_ = std::make_unique<AudioDebugger>();
     audio_processor_->Initialize(codec);
     audio_processor_->OnOutput([this](std::vector<int16_t>&& data) {
         {
@@ -884,6 +886,12 @@ bool Application::ReadAudio(std::vector<int16_t>& data, int sample_rate, int sam
             return false;
         }
     }
+    
+    // 音频调试：发送原始音频数据
+    if (audio_debugger_) {
+        audio_debugger_->Feed(data);
+    }
+    
     return true;
 }
 
diff --git a/main/application.h b/main/application.h
index fd138447..3a9b6fd1 100644
--- a/main/application.h
+++ b/main/application.h
@@ -22,6 +22,7 @@
 #include "background_task.h"
 #include "audio_processor.h"
 #include "wake_word.h"
+#include "audio_debugger.h"
 
 #define SCHEDULE_EVENT (1 << 0)
 #define SEND_AUDIO_EVENT (1 << 1)
@@ -86,6 +87,7 @@ private:
 
     std::unique_ptr<WakeWord> wake_word_;
     std::unique_ptr<AudioProcessor> audio_processor_;
+    std::unique_ptr<AudioDebugger> audio_debugger_;
     Ota ota_;
     std::mutex mutex_;
     std::list<std::function<void()>> main_tasks_;
diff --git a/main/audio_codecs/audio_codec.h b/main/audio_codecs/audio_codec.h
index beb71a6d..fb561061 100644
--- a/main/audio_codecs/audio_codec.h
+++ b/main/audio_codecs/audio_codec.h
@@ -13,7 +13,7 @@
 
 #define AUDIO_CODEC_DMA_DESC_NUM 6
 #define AUDIO_CODEC_DMA_FRAME_NUM 240
-#define AUDIO_CODEC_DEFAULT_MIC_GAIN 36.0
+#define AUDIO_CODEC_DEFAULT_MIC_GAIN 30.0
 
 class AudioCodec {
 public:
diff --git a/main/audio_codecs/box_audio_codec.cc b/main/audio_codecs/box_audio_codec.cc
index 58e0270d..acf8f2a8 100644
--- a/main/audio_codecs/box_audio_codec.cc
+++ b/main/audio_codecs/box_audio_codec.cc
@@ -199,7 +199,7 @@ void BoxAudioCodec::EnableInput(bool enable) {
             fs.channel_mask |= ESP_CODEC_DEV_MAKE_CHANNEL_MASK(1);
         }
         ESP_ERROR_CHECK(esp_codec_dev_open(input_dev_, &fs));
-        ESP_ERROR_CHECK(esp_codec_dev_set_in_channel_gain(input_dev_, ESP_CODEC_DEV_MAKE_CHANNEL_MASK(0), 36.0));
+        ESP_ERROR_CHECK(esp_codec_dev_set_in_channel_gain(input_dev_, ESP_CODEC_DEV_MAKE_CHANNEL_MASK(0), AUDIO_CODEC_DEFAULT_MIC_GAIN));
     } else {
         ESP_ERROR_CHECK(esp_codec_dev_close(input_dev_));
     }
diff --git a/main/audio_processing/afe_wake_word.cc b/main/audio_processing/afe_wake_word.cc
index af7bfa06..77955970 100644
--- a/main/audio_processing/afe_wake_word.cc
+++ b/main/audio_processing/afe_wake_word.cc
@@ -155,18 +155,19 @@ void AfeWakeWord::EncodeWakeWordData() {
             auto encoder = std::make_unique<OpusEncoderWrapper>(16000, 1, OPUS_FRAME_DURATION_MS);
             encoder->SetComplexity(0); // 0 is the fastest
 
+            int packets = 0;
             for (auto& pcm: this_->wake_word_pcm_) {
                 encoder->Encode(std::move(pcm), [this_](std::vector<uint8_t>&& opus) {
                     std::lock_guard<std::mutex> lock(this_->wake_word_mutex_);
                     this_->wake_word_opus_.emplace_back(std::move(opus));
                     this_->wake_word_cv_.notify_all();
                 });
+                packets++;
             }
             this_->wake_word_pcm_.clear();
 
             auto end_time = esp_timer_get_time();
-            ESP_LOGI(TAG, "Encode wake word opus %u packets in %lld ms",
-                this_->wake_word_opus_.size(), (end_time - start_time) / 1000);
+            ESP_LOGI(TAG, "Encode wake word opus %d packets in %ld ms", packets, (long)((end_time - start_time) / 1000));
 
             std::lock_guard<std::mutex> lock(this_->wake_word_mutex_);
             this_->wake_word_opus_.push_back(std::vector<uint8_t>());
diff --git a/main/audio_processing/audio_debugger.cc b/main/audio_processing/audio_debugger.cc
new file mode 100644
index 00000000..216b2258
--- /dev/null
+++ b/main/audio_processing/audio_debugger.cc
@@ -0,0 +1,64 @@
+#include "audio_debugger.h"
+#include "sdkconfig.h"
+
+#if CONFIG_USE_AUDIO_DEBUGGER
+#include <esp_log.h>
+#include <arpa/inet.h>
+#include <unistd.h>
+#include <errno.h>
+#include <cstring>
+#include <string>
+#endif
+
+#define TAG "AudioDebugger"
+
+
+AudioDebugger::AudioDebugger() {
+#if CONFIG_USE_AUDIO_DEBUGGER
+    udp_sockfd_ = socket(AF_INET, SOCK_DGRAM, 0);
+    if (udp_sockfd_ >= 0) {
+        // 解析配置的服务器地址 "IP:PORT"
+        std::string server_addr = CONFIG_AUDIO_DEBUG_UDP_SERVER;
+        size_t colon_pos = server_addr.find(':');
+        
+        if (colon_pos != std::string::npos) {
+            std::string ip = server_addr.substr(0, colon_pos);
+            int port = std::stoi(server_addr.substr(colon_pos + 1));
+            
+            memset(&udp_server_addr_, 0, sizeof(udp_server_addr_));
+            udp_server_addr_.sin_family = AF_INET;
+            udp_server_addr_.sin_port = htons(port);
+            inet_pton(AF_INET, ip.c_str(), &udp_server_addr_.sin_addr);
+            
+            ESP_LOGI(TAG, "Initialized server address: %s", CONFIG_AUDIO_DEBUG_UDP_SERVER);
+        } else {
+            ESP_LOGW(TAG, "Invalid server address: %s, should be IP:PORT", CONFIG_AUDIO_DEBUG_UDP_SERVER);
+            close(udp_sockfd_);
+            udp_sockfd_ = -1;
+        }
+    } else {
+        ESP_LOGW(TAG, "Failed to create UDP socket: %d", errno);
+    }
+#endif
+}
+
+AudioDebugger::~AudioDebugger() {
+    if (udp_sockfd_ >= 0) {
+        close(udp_sockfd_);
+        ESP_LOGI(TAG, "Closed UDP socket");
+    }
+}
+
+void AudioDebugger::Feed(const std::vector<int16_t>& data) {
+    if (udp_sockfd_ >= 0) {
+        ssize_t sent = sendto(udp_sockfd_, data.data(), data.size() * sizeof(int16_t), 0,
+                             (struct sockaddr*)&udp_server_addr_, sizeof(udp_server_addr_));
+        if (sent < 0) {
+            ESP_LOGW(TAG, "Failed to send audio data to %s: %d", CONFIG_AUDIO_DEBUG_UDP_SERVER, errno);
+        } else {
+            ESP_LOGD(TAG, "Sent %d bytes audio data to %s", sent, CONFIG_AUDIO_DEBUG_UDP_SERVER);
+        }
+    }
+}
+
+ 
\ No newline at end of file
diff --git a/main/audio_processing/audio_debugger.h b/main/audio_processing/audio_debugger.h
new file mode 100644
index 00000000..a81336cb
--- /dev/null
+++ b/main/audio_processing/audio_debugger.h
@@ -0,0 +1,22 @@
+#ifndef AUDIO_DEBUGGER_H
+#define AUDIO_DEBUGGER_H
+
+#include <vector>
+#include <cstdint>
+
+#include <sys/socket.h>
+#include <netinet/in.h>
+
+class AudioDebugger {
+public:
+    AudioDebugger();
+    ~AudioDebugger();
+
+    void Feed(const std::vector<int16_t>& data);
+
+private:
+    int udp_sockfd_ = -1;
+    struct sockaddr_in udp_server_addr_;
+};
+
+#endif 
\ No newline at end of file
diff --git a/main/protocols/protocol.cc b/main/protocols/protocol.cc
index d5c1cb5e..1d915be8 100644
--- a/main/protocols/protocol.cc
+++ b/main/protocols/protocol.cc
@@ -126,7 +126,7 @@ bool Protocol::IsTimeout() const {
     auto duration = std::chrono::duration_cast<std::chrono::seconds>(now - last_incoming_time_);
     bool timeout = duration.count() > kTimeoutSeconds;
     if (timeout) {
-        ESP_LOGE(TAG, "Channel timeout %lld seconds", duration.count());
+        ESP_LOGE(TAG, "Channel timeout %ld seconds", (long)duration.count());
     }
     return timeout;
 }
diff --git a/scripts/audio_debug_server.py b/scripts/audio_debug_server.py
new file mode 100644
index 00000000..872c4905
--- /dev/null
+++ b/scripts/audio_debug_server.py
@@ -0,0 +1,54 @@
+import socket
+import wave
+import argparse
+
+
+'''
+  Create a UDP socket and bind it to the server's IP:8000.
+  Listen for incoming messages and print them to the console.
+  Save the audio to a WAV file.
+'''
+def main(samplerate, channels):
+    # Create a UDP socket
+    server_socket = socket.socket(socket.AF_INET, socket.SOCK_DGRAM)
+    server_socket.bind(('0.0.0.0', 8000))
+
+    # Create WAV file with parameters
+    filename = f"{samplerate}_{channels}.wav"
+    wav_file = wave.open(filename, "wb")
+    wav_file.setnchannels(channels)     # channels parameter
+    wav_file.setsampwidth(2)            # 2 bytes per sample (16-bit)
+    wav_file.setframerate(samplerate)   # samplerate parameter
+
+    print(f"Start saving audio from 0.0.0.0:8000 to {filename}...")
+
+    try:
+        while True:
+            # Receive a message from the client
+            message, address = server_socket.recvfrom(8000)
+            
+            # Write PCM data to WAV file
+            wav_file.writeframes(message)
+
+            # Print length of the message
+            print(f"Received {len(message)} bytes from {address}")
+    
+    except KeyboardInterrupt:
+        print("\nStopping recording...")
+    
+    finally:
+        # Close files and socket
+        wav_file.close()
+        server_socket.close()
+        print(f"WAV file '{filename}' saved successfully")
+
+
+if __name__ == "__main__":
+    parser = argparse.ArgumentParser(description='UDP音频数据接收器，保存为WAV文件')
+    parser.add_argument('--samplerate', '-s', type=int, default=16000, 
+                        help='采样率 (默认: 16000)')
+    parser.add_argument('--channels', '-c', type=int, default=2, 
+                        help='声道数 (默认: 2)')
+    
+    args = parser.parse_args()
+    main(args.samplerate, args.channels)