diff --git a/Record/ProcessFile.cpp b/Record/ProcessFile.cpp
index b4c0d01..883b12d 100644
--- a/Record/ProcessFile.cpp
+++ b/Record/ProcessFile.cpp
@@ -6,6 +6,7 @@
 #include "api/audio/echo_canceller3_factory.h"
 #include "main.h"
 #include "modules/audio_processing/aec3/echo_canceller3.h"
+#include "modules/audio_processing/ns/noise_suppressor.h"
 #include <boost/asio/post.hpp>
 #include <sstream>
 
@@ -19,6 +20,10 @@ public:
         linearOutputBuffer = std::make_unique<webrtc::AudioBuffer>(sampleRate, channels, sampleRate, channels, sampleRate, channels);
 
         // RKAP_3A_Init(&m_vqe, AEC_TX_TYPE);
+        webrtc::NsConfig config;
+        config.target_level = webrtc::NsConfig::SuppressionLevel::k12dB;
+        noiseSuppressor = std::make_unique<webrtc::NoiseSuppressor>(config, sampleRate, channels);
+        noiseSuppressorBuffer = std::make_unique<webrtc::AudioBuffer>(sampleRate, channels, sampleRate, channels, sampleRate, channels);
     }
 
     std::unique_ptr<webrtc::EchoControl> echoCanceller;
@@ -26,6 +31,9 @@ public:
     std::unique_ptr<webrtc::AudioBuffer> farendBuffer;
     std::unique_ptr<webrtc::AudioBuffer> linearOutputBuffer;
 
+    std::unique_ptr<webrtc::NoiseSuppressor> noiseSuppressor;
+    std::unique_ptr<webrtc::AudioBuffer> noiseSuppressorBuffer;
+
     // RKAP_AEC_State m_vqe;
 };
 
@@ -54,6 +62,10 @@ void ProcessFileTask::run() {
     m_outFilename = oss.str();
     m_ofs = std::make_shared<std::ofstream>(m_outFilename, std::ofstream::binary);
 
+    oss.str("");
+    oss << DumpPath << "/ns_" << dspToString(m_dsp) << "_16k.pcm";
+    m_nsOfs = std::make_shared<std::ofstream>(oss.str(), std::ofstream::binary);
+
     // m_speakerIfs = std::make_shared<std::ifstream>("/sdcard/speaker_8k.pcm", std::ifstream::binary);
     // m_micIfs = std::make_shared<std::ifstream>("/sdcard/micin_8k.pcm", std::ifstream::binary);
     // m_ofs = std::make_shared<std::ofstream>("/sdcard/out_8k.pcm", std::ofstream::binary);
@@ -75,23 +87,32 @@ void ProcessFileTask::process() {
     if (m_speakerIfs && m_micIfs && *m_speakerIfs && *m_micIfs) {
         char farendBuffer[sizeof(int16_t) * 16000 / 1000 * 10] = {0};
         char nearendBuffer[sizeof(int16_t) * 16000 / 1000 * 10] = {0};
+        char noiseSuppressorOutBuffer[sizeof(int16_t) * 16000 / 1000 * 10] = {0};
         char outBuffer[sizeof(int16_t) * 16000 / 1000 * 10] = {0};
+        webrtc::StreamConfig config(16000, 1); // 单声道
+
         // char farendBuffer[sizeof(int16_t) * 8000 / 1000 * 10] = {0};
         // char nearendBuffer[sizeof(int16_t) * 8000 / 1000 * 10] = {0};
         // char outBuffer[sizeof(int16_t) * 8000 / 1000 * 10] = {0};
 
         m_speakerIfs->read(farendBuffer, sizeof(farendBuffer));
         m_micIfs->read(nearendBuffer, sizeof(nearendBuffer));
+        m_d->nearendBuffer->CopyFrom(reinterpret_cast<const int16_t *>(nearendBuffer), config);
         if (m_dsp == Speex) {
             m_speex->echoPlayback(reinterpret_cast<const int16_t *>(farendBuffer));
             m_speex->echoCapture(reinterpret_cast<const int16_t *>(nearendBuffer), reinterpret_cast<int16_t *>(outBuffer));
         } else if (m_dsp == AecMobile) {
+            m_d->noiseSuppressor->Analyze(*m_d->nearendBuffer);
+            m_d->noiseSuppressor->Process(m_d->nearendBuffer.get());
+            m_d->nearendBuffer->CopyTo(config, reinterpret_cast<int16_t *>(noiseSuppressorOutBuffer));
+            m_nsOfs->write(noiseSuppressorOutBuffer, sizeof(noiseSuppressorOutBuffer));
+            // LOG(info) << " " << m_d->noiseSuppressorBuffer->num_frames() << " " << m_d->nearendBuffer->num_frames();
+
             m_webRtcAecm->echoPlayback(reinterpret_cast<const int16_t *>(farendBuffer), sizeof(farendBuffer) / 2);
-            m_webRtcAecm->echoCancellation(reinterpret_cast<int16_t *>(nearendBuffer), nullptr, reinterpret_cast<int16_t *>(outBuffer),
+            m_webRtcAecm->echoCancellation(reinterpret_cast<int16_t *>(nearendBuffer),
+                                           reinterpret_cast<int16_t *>(noiseSuppressorOutBuffer), reinterpret_cast<int16_t *>(outBuffer),
                                            sizeof(farendBuffer) / 2);
         } else if (m_dsp == Aec3) {
-            webrtc::StreamConfig config(16000, 1); // 单声道
-            m_d->nearendBuffer->CopyFrom(reinterpret_cast<const int16_t *>(nearendBuffer), config);
 
             m_d->farendBuffer->CopyFrom(reinterpret_cast<const int16_t *>(farendBuffer), config);
 
diff --git a/Record/main.h b/Record/main.h
index 7745946..685c619 100644
--- a/Record/main.h
+++ b/Record/main.h
@@ -96,6 +96,7 @@ private:
     std::shared_ptr<std::ifstream> m_speakerIfs;
     std::shared_ptr<std::ifstream> m_micIfs;
     std::shared_ptr<std::ofstream> m_ofs;
+    std::shared_ptr<std::ofstream> m_nsOfs; // 降噪后的文件
     std::string m_outFilename;
     std::chrono::system_clock::time_point m_begin;
 };