diff --git a/frameworks/native/audioadapter/src/pulse_audio_service_adapter_impl.cpp b/frameworks/native/audioadapter/src/pulse_audio_service_adapter_impl.cpp
index a7cabeb9a6373a136ae07fb737479d742b1b7548..6af9d39ba5459dfda12f184a1f46ed1f4e1a18f1 100644
--- a/frameworks/native/audioadapter/src/pulse_audio_service_adapter_impl.cpp
+++ b/frameworks/native/audioadapter/src/pulse_audio_service_adapter_impl.cpp
@@ -878,7 +878,7 @@ void PulseAudioServiceAdapterImpl::PaGetAllSourceOutputsCb(pa_context *c, const
     CHECK_AND_RETURN_LOG(i->proplist != nullptr,
         "Invalid Proplist for source output (%{public}d).", i->index);
 
-    uint32_t sessionID = 0;
+    int32_t sessionID = 0;
     const char *sessionCStr = pa_proplist_gets(i->proplist, "stream.sessionID");
     if (sessionCStr != nullptr) {
         std::stringstream sessionStr;
diff --git a/frameworks/native/audioutils/include/audio_utils.h b/frameworks/native/audioutils/include/audio_utils.h
index cd85b41411a341735db78d769f300e3c77eca034..08dadbe045aacc491b95029ec4a2ff0434ba44e1 100644
--- a/frameworks/native/audioutils/include/audio_utils.h
+++ b/frameworks/native/audioutils/include/audio_utils.h
@@ -26,7 +26,7 @@
 
 #define AUDIO_MS_PER_SECOND 1000
 #define AUDIO_US_PER_SECOND 1000000
-#define AUDIO_NS_PER_SECOND ((int64_t)1000000000)
+#define AUDIO_NS_PER_SECOND ((uint64_t)1000000000)
 
 #define FLOAT_EPS 1e-9f
 #define OFFSET_BIT_24 3
diff --git a/frameworks/native/audioutils/src/audio_channel_blend.cpp b/frameworks/native/audioutils/src/audio_channel_blend.cpp
index 99c6e94389dd0c8882b59fe7e9497d39184a458c..bf6e370bab2e789dafe4a31abc9fe46ba8353e59 100644
--- a/frameworks/native/audioutils/src/audio_channel_blend.cpp
+++ b/frameworks/native/audioutils/src/audio_channel_blend.cpp
@@ -69,7 +69,7 @@ void AudioBlend::ProcessWithBlendMode(T *buffer, size_t bufferSize)
         return;
     }
 
-    int frameCount = 0;
+    uint32_t frameCount = 0;
     frameCount = bufferSize / (channels_ * (format_ + 1));
     switch (blendMode_) {
         case MODE_BLEND_LR:
@@ -107,7 +107,7 @@ void AudioBlend::BlendLR(int24_t& left, int24_t& right)
 template <typename T>
 void AudioBlend::ProcessBlendLRModeWithFormat(T *buffer, size_t count, AudioChannel channel)
 {
-    for (int i = count; i > 0; i--) {
+    for (uint32_t i = count; i > 0; i--) {
         switch (channel) {
             case CHANNEL_8:
                 BlendLR(buffer[CHANNEL_SEVEN], buffer[CHANNEL_EIGHT]);
diff --git a/frameworks/native/audioutils/src/audio_utils.cpp b/frameworks/native/audioutils/src/audio_utils.cpp
index 205676d17eea8a44a0c933e9bc00ac7af543c6d7..4d470f0c651ceba05800531efd7c637aa1b267a0 100644
--- a/frameworks/native/audioutils/src/audio_utils.cpp
+++ b/frameworks/native/audioutils/src/audio_utils.cpp
@@ -413,7 +413,7 @@ float UpdateMaxAmplitude(ConvertHdiFormat adapterFormat, char *frame, uint64_t r
 float CalculateMaxAmplitudeForPCM8Bit(int8_t *frame, uint64_t nSamples)
 {
     int curMaxAmplitude = 0;
-    for (int i = nSamples; i > 0; --i) {
+    for (uint32_t i = nSamples; i > 0; --i) {
         int8_t value = *frame++;
         if (value < 0) {
             value = -value;
@@ -428,7 +428,7 @@ float CalculateMaxAmplitudeForPCM8Bit(int8_t *frame, uint64_t nSamples)
 float CalculateMaxAmplitudeForPCM16Bit(int16_t *frame, uint64_t nSamples)
 {
     int curMaxAmplitude = 0;
-    for (int i = nSamples; i > 0; --i) {
+    for (uint32_t i = nSamples; i > 0; --i) {
         int16_t value = *frame++;
         if (value < 0) {
             value = -value;
@@ -443,7 +443,7 @@ float CalculateMaxAmplitudeForPCM16Bit(int16_t *frame, uint64_t nSamples)
 float CalculateMaxAmplitudeForPCM24Bit(char *frame, uint64_t nSamples)
 {
     int curMaxAmplitude = 0;
-    for (int i = 0; i < nSamples; ++i) {
+    for (uint32_t i = 0; i < nSamples; ++i) {
         char *curPos = frame + (i * 3); // 3 bytes
         int curValue = 0;
         for (int j = 0; j < 3; ++j) { // 3 bytes
@@ -462,7 +462,7 @@ float CalculateMaxAmplitudeForPCM24Bit(char *frame, uint64_t nSamples)
 float CalculateMaxAmplitudeForPCM32Bit(int32_t *frame, uint64_t nSamples)
 {
     int curMaxAmplitude = 0;
-    for (int i = nSamples; i > 0; --i) {
+    for (uint32_t i = nSamples; i > 0; --i) {
         int32_t value = *frame++;
         if (value < 0) {
             value = -value;
@@ -687,7 +687,7 @@ bool SignalDetectAgent::CheckAudioData(uint8_t *buffer, size_t bufferLen)
 bool SignalDetectAgent::DetectSignalData(int32_t *buffer, size_t bufferLen)
 {
     std::string curTime = GetTime();
-    int32_t rightZeroSignal = 0;
+    uint32_t rightZeroSignal = 0;
     int32_t currentPeakIndex = -1;
     int32_t currentPeakSignal = SHRT_MIN;
     int32_t tempMax;
diff --git a/frameworks/native/hdiadapter/devicemanager/src/audio_device_manager_impl.cpp b/frameworks/native/hdiadapter/devicemanager/src/audio_device_manager_impl.cpp
index 09395445695376a4e8a13aef1896acdcfebdd03f..dad208c10f11a4dd8440da4ec0b7981e1d5df461 100644
--- a/frameworks/native/hdiadapter/devicemanager/src/audio_device_manager_impl.cpp
+++ b/frameworks/native/hdiadapter/devicemanager/src/audio_device_manager_impl.cpp
@@ -120,7 +120,7 @@ struct AudioAdapterDescriptor *AudioDeviceManagerImpl::GetTargetAdapterDesc(cons
     CHECK_AND_RETURN_RET_LOG(ret == SUCCESS, nullptr,
         "Get all adapters fail, audioMgrType %{public}d, ret %{public}d.", audioMgrType_, ret);
 
-    int32_t targetIdx = INVALID_INDEX;
+    uint32_t targetIdx = INVALID_INDEX;
     for (uint32_t index = 0; index < descriptors_.size(); index++) {
         struct AudioAdapterDescriptor desc = descriptors_[index];
         if (desc.adapterName.c_str() == nullptr) {
diff --git a/frameworks/native/hdiadapter/sink/fast/fast_audio_renderer_sink.cpp b/frameworks/native/hdiadapter/sink/fast/fast_audio_renderer_sink.cpp
index 0779b2a76487fc41290d449bbeb80992899c4aca..b4c808f4ca49a938b8df50a0e27d13c352c91090 100644
--- a/frameworks/native/hdiadapter/sink/fast/fast_audio_renderer_sink.cpp
+++ b/frameworks/native/hdiadapter/sink/fast/fast_audio_renderer_sink.cpp
@@ -346,9 +346,9 @@ void FastAudioRendererSinkInner::ReleaseMmapBuffer()
 
 int32_t FastAudioRendererSinkInner::PrepareMmapBuffer()
 {
-    int32_t totalBifferInMs = 40; // 5 * (6 + 2 * (1)) = 40ms, the buffer size, not latency.
+    uint32_t totalBifferInMs = 40; // 5 * (6 + 2 * (1)) = 40ms, the buffer size, not latency.
     frameSizeInByte_ = PcmFormatToBits(attr_.format) * attr_.channel / PCM_8_BIT;
-    int32_t reqBufferFrameSize = totalBifferInMs * (attr_.sampleRate / 1000);
+    uint32_t reqBufferFrameSize = totalBifferInMs * (attr_.sampleRate / 1000);
 
     struct AudioMmapBufferDescriptor desc = {0};
     int32_t ret = audioRender_->ReqMmapBuffer(audioRender_, reqBufferFrameSize, &desc);
diff --git a/frameworks/native/hdiadapter/sink/offload/offload_audio_renderer_sink.cpp b/frameworks/native/hdiadapter/sink/offload/offload_audio_renderer_sink.cpp
index 543af76636a4679e4afe59b7bdfd6db2dc4e0163..29002c6ab7f34a5926a4e4516a3ce740b5c5e540 100644
--- a/frameworks/native/hdiadapter/sink/offload/offload_audio_renderer_sink.cpp
+++ b/frameworks/native/hdiadapter/sink/offload/offload_audio_renderer_sink.cpp
@@ -55,9 +55,9 @@ const uint32_t STEREO_CHANNEL_COUNT = 2;
 #ifdef FEATURE_POWER_MANAGER
 constexpr int32_t RUNNINGLOCK_LOCK_TIMEOUTMS_LASTING = -1;
 #endif
-const int64_t SECOND_TO_NANOSECOND = 1000000000;
-const int64_t SECOND_TO_MICROSECOND = 1000000;
-const int64_t SECOND_TO_MILLISECOND = 1000;
+const uint64_t SECOND_TO_NANOSECOND = 1000000000;
+const uint64_t SECOND_TO_MICROSECOND = 1000000;
+const uint64_t SECOND_TO_MILLISECOND = 1000;
 const uint32_t BIT_IN_BYTE = 8;
 const uint16_t GET_MAX_AMPLITUDE_FRAMES_THRESHOLD = 10;
 }
@@ -841,7 +841,7 @@ int32_t OffloadAudioRendererSinkInner::SetBufferSize(uint32_t sizeMs)
     int32_t ret;
 
     // bytewidth is 4
-    uint32_t size = (int64_t) sizeMs * AUDIO_SAMPLE_RATE_48K * 4 * STEREO_CHANNEL_COUNT / SECOND_TO_MILLISECOND;
+    uint32_t size = (uint64_t) sizeMs * AUDIO_SAMPLE_RATE_48K * 4 * STEREO_CHANNEL_COUNT / SECOND_TO_MILLISECOND;
     CHECK_AND_RETURN_RET_LOG(audioRender_ != nullptr, ERR_INVALID_HANDLE,
         "failed audio render null");
 
@@ -924,7 +924,7 @@ void OffloadAudioRendererSinkInner::CheckLatencySignal(uint8_t *data, size_t len
         return;
     }
     CHECK_AND_RETURN_LOG(signalDetectAgent_ != nullptr, "LatencyMeas signalDetectAgent_ is nullptr");
-    int32_t byteSize = GetFormatByteSize(attr_.format);
+    uint32_t byteSize = GetFormatByteSize(attr_.format);
     size_t newlyCheckedTime = len / (attr_.sampleRate / MILLISECOND_PER_SECOND) /
         (byteSize * sizeof(uint8_t) * attr_.channel);
     detectedTime_ += newlyCheckedTime;
diff --git a/frameworks/native/hdiadapter/sink/primary/audio_renderer_sink.cpp b/frameworks/native/hdiadapter/sink/primary/audio_renderer_sink.cpp
index 125561f82865cf651f69a1d0c09d928d314a29e7..c9528cdac2e3648db9f3e554d387b14d8bdcc72e 100644
--- a/frameworks/native/hdiadapter/sink/primary/audio_renderer_sink.cpp
+++ b/frameworks/native/hdiadapter/sink/primary/audio_renderer_sink.cpp
@@ -1208,7 +1208,7 @@ int32_t AudioRendererSinkInner::UpdateDPAttrs(const std::string &usbInfoStr)
     if (attr_.channel <= 0 || attr_.sampleRate <= 0) {
         AUDIO_ERR_LOG("check attr failed channel[%{public}d] sampleRate[%{public}d]", attr_.channel, attr_.sampleRate);
     }
-    int32_t formatByte = stoi(bufferSize) * BUFFER_CALC_1000MS / BUFFER_CALC_20MS / attr_.channel / attr_.sampleRate;
+    uint32_t formatByte = stoi(bufferSize) * BUFFER_CALC_1000MS / BUFFER_CALC_20MS / attr_.channel / attr_.sampleRate;
     attr_.format = static_cast<HdiAdapterFormat>(ConvertByteToAudioFormat(formatByte));
 
     AUDIO_DEBUG_LOG("UpdateDPAttrs sampleRate %{public}d, format: %{public}d,channelCount: %{public}d",
@@ -1325,7 +1325,7 @@ void AudioRendererSinkInner::CheckLatencySignal(uint8_t *data, size_t len)
         return;
     }
     CHECK_AND_RETURN_LOG(signalDetectAgent_ != nullptr, "LatencyMeas signalDetectAgent_ is nullptr");
-    int32_t byteSize = GetFormatByteSize(attr_.format);
+    uint32_t byteSize = GetFormatByteSize(attr_.format);
     size_t newlyCheckedTime = len / (attr_.sampleRate / MILLISECOND_PER_SECOND) /
         (byteSize * sizeof(uint8_t) * attr_.channel);
     detectedTime_ += newlyCheckedTime;
diff --git a/frameworks/native/hdiadapter/source/fast/fast_audio_capturer_source.cpp b/frameworks/native/hdiadapter/source/fast/fast_audio_capturer_source.cpp
index 31140984911a54ee6238dbb5a6b250a7f6e16c56..70d8932bc8ec8a018dff2a1b6b428c2ad818dabb 100644
--- a/frameworks/native/hdiadapter/source/fast/fast_audio_capturer_source.cpp
+++ b/frameworks/native/hdiadapter/source/fast/fast_audio_capturer_source.cpp
@@ -338,9 +338,9 @@ int32_t FastAudioCapturerSourceInner::GetMmapHandlePosition(uint64_t &frames, in
 
 int32_t FastAudioCapturerSourceInner::PrepareMmapBuffer()
 {
-    int32_t totalBifferInMs = 40; // 5 * (6 + 2 * (1)) = 40ms, the buffer size, not latency.
+    uint32_t totalBifferInMs = 40; // 5 * (6 + 2 * (1)) = 40ms, the buffer size, not latency.
     uint32_t frameSizeInByte = PcmFormatToBits(attr_.format) * attr_.channel / PCM_8_BIT;
-    int32_t reqBufferFrameSize = totalBifferInMs * (attr_.sampleRate / 1000);
+    uint32_t reqBufferFrameSize = totalBifferInMs * (attr_.sampleRate / 1000);
 
     struct AudioMmapBufferDescriptor desc = {0};
     int32_t ret = audioCapture_->ReqMmapBuffer(audioCapture_, reqBufferFrameSize, &desc);
diff --git a/frameworks/native/ohaudio/OHAudioRoutingManager.cpp b/frameworks/native/ohaudio/OHAudioRoutingManager.cpp
index 78d33176402790d39a6f18b8ffa00029b9a3e686..93883355841fd978de08073c9ec6199e6fee252f 100644
--- a/frameworks/native/ohaudio/OHAudioRoutingManager.cpp
+++ b/frameworks/native/ohaudio/OHAudioRoutingManager.cpp
@@ -97,7 +97,7 @@ OH_AudioCommon_Result OH_AudioRoutingManager_ReleaseDevices(
     if (audioDeviceDescriptorArray == nullptr) {
         return AUDIOCOMMON_RESULT_ERROR_INVALID_PARAM;
     }
-    for (int32_t index = 0; index < audioDeviceDescriptorArray->size; index++) {
+    for (uint32_t index = 0; index < audioDeviceDescriptorArray->size; index++) {
         OHAudioDeviceDescriptor* ohAudioDeviceDescriptor =
             (OHAudioDeviceDescriptor*)audioDeviceDescriptorArray->descriptors[index];
         delete ohAudioDeviceDescriptor;
@@ -116,7 +116,7 @@ namespace AudioStandard {
 void DestroyAudioDeviceDescriptor(OH_AudioDeviceDescriptorArray *array)
 {
     if (array) {
-        for (int32_t index = 0; index < array->size; index++) {
+        for (uint32_t index = 0; index < array->size; index++) {
             OHAudioDeviceDescriptor* ohAudioDeviceDescriptor = (OHAudioDeviceDescriptor*)array->descriptors[index];
             delete ohAudioDeviceDescriptor;
             array->descriptors[index] = nullptr;
diff --git a/services/audio_policy/client/src/audio_policy_proxy.cpp b/services/audio_policy/client/src/audio_policy_proxy.cpp
index 7ef89a138839012bfeb7f36bbeb13d45259044ec..ce4bbac88a3452be342de71a1f9dd50ce13e7f83 100644
--- a/services/audio_policy/client/src/audio_policy_proxy.cpp
+++ b/services/audio_policy/client/src/audio_policy_proxy.cpp
@@ -513,7 +513,7 @@ int32_t AudioPolicyProxy::SelectOutputDevice(sptr<AudioRendererFilter> audioRend
     bool tmp = audioRendererFilter->Marshalling(data);
     CHECK_AND_RETURN_RET_LOG(tmp, -1, "AudioRendererFilter Marshalling() failed");
 
-    int size = audioDeviceDescriptors.size();
+    uint32_t size = audioDeviceDescriptors.size();
     int validSize = 20; // Use 20 as limit.
     CHECK_AND_RETURN_RET_LOG(size > 0 && size <= validSize, -1, "SelectOutputDevice get invalid device size.");
     data.WriteInt32(size);
@@ -557,7 +557,7 @@ int32_t AudioPolicyProxy::SelectInputDevice(sptr<AudioCapturerFilter> audioCaptu
     bool tmp = audioCapturerFilter->Marshalling(data);
     CHECK_AND_RETURN_RET_LOG(tmp, -1, "AudioCapturerFilter Marshalling() failed");
 
-    int size = audioDeviceDescriptors.size();
+    uint32_t size = audioDeviceDescriptors.size();
     int validSize = 20; // Use 20 as limit.
     CHECK_AND_RETURN_RET_LOG(size > 0 && size <= validSize, -1, "SelectOutputDevice get invalid device size.");
     data.WriteInt32(size);
diff --git a/services/audio_policy/server/include/service/audio_policy_service.h b/services/audio_policy/server/include/service/audio_policy_service.h
index 6d4780b5ca0c580e545c12215a5a37f5ee403d32..1732e20f8d54c72353252e8ffb8c7cdd81376806 100644
--- a/services/audio_policy/server/include/service/audio_policy_service.h
+++ b/services/audio_policy/server/include/service/audio_policy_service.h
@@ -710,7 +710,7 @@ private:
 
     void ClearScoDeviceSuspendState(string macAddress = "");
 
-    int32_t OpenPortAndInsertIOHandle(const std::string &moduleName, const AudioModuleInfo &moduleInfo);
+    AudioIOHandle OpenPortAndInsertIOHandle(const std::string &moduleName, const AudioModuleInfo &moduleInfo);
 
     int32_t ClosePortAndEraseIOHandle(const std::string &moduleName);
 
diff --git a/services/audio_policy/server/src/audio_policy_manager_stub.cpp b/services/audio_policy/server/src/audio_policy_manager_stub.cpp
index 3a757ca81d9cf68203e07f4ff176d706ac1041e4..77927416af9ac268289b8b8a2798b641b7c915db 100644
--- a/services/audio_policy/server/src/audio_policy_manager_stub.cpp
+++ b/services/audio_policy/server/src/audio_policy_manager_stub.cpp
@@ -261,7 +261,7 @@ void AudioPolicyManagerStub::GetPreferredInputDeviceDescriptorsInternal(MessageP
     AudioCapturerInfo captureInfo;
     captureInfo.Unmarshalling(data);
     std::vector<sptr<AudioDeviceDescriptor>> devices = GetPreferredInputDeviceDescriptors(captureInfo);
-    size_t size = static_cast<int32_t>(devices.size());
+    int32_t size = static_cast<int32_t>(devices.size());
     reply.WriteInt32(size);
     for (size_t i = 0; i < size; i++) {
         devices[i]->Marshalling(reply);
@@ -463,7 +463,7 @@ void AudioPolicyManagerStub::CheckRecordingCreateInternal(MessageParcel &data, M
 {
     uint32_t appTokenId = data.ReadUint32();
     uint64_t appFullTokenId = data.ReadUint64();
-    uint32_t appUid = data.ReadInt32();
+    int32_t appUid = data.ReadInt32();
     SourceType sourceType = static_cast<SourceType> (data.ReadInt32());
     bool ret = CheckRecordingCreate(appTokenId, appFullTokenId, appUid, sourceType);
     reply.WriteBool(ret);
@@ -647,7 +647,7 @@ void AudioPolicyManagerStub::GetMaxRendererInstancesInternal(MessageParcel &data
 static void PreprocessMode(SupportedEffectConfig &supportedEffectConfig, MessageParcel &reply, int32_t i, int32_t j)
 {
     reply.WriteString(supportedEffectConfig.preProcessNew.stream[i].streamEffectMode[j].mode);
-    int32_t countDev = supportedEffectConfig.preProcessNew.stream[i].streamEffectMode[j].devicePort.size();
+    uint32_t countDev = supportedEffectConfig.preProcessNew.stream[i].streamEffectMode[j].devicePort.size();
     reply.WriteInt32(countDev);
     if (countDev > 0) {
         for (int32_t k = 0; k < countDev; k++) {
@@ -659,7 +659,7 @@ static void PreprocessMode(SupportedEffectConfig &supportedEffectConfig, Message
 static void PreprocessProcess(SupportedEffectConfig &supportedEffectConfig, MessageParcel &reply, int32_t i)
 {
     reply.WriteString(supportedEffectConfig.preProcessNew.stream[i].scene);
-    int32_t countMode = supportedEffectConfig.preProcessNew.stream[i].streamEffectMode.size();
+    uint32_t countMode = supportedEffectConfig.preProcessNew.stream[i].streamEffectMode.size();
     reply.WriteInt32(countMode);
     if (countMode > 0) {
         for (int32_t j = 0; j < countMode; j++) {
@@ -670,7 +670,7 @@ static void PreprocessProcess(SupportedEffectConfig &supportedEffectConfig, Mess
 static void PostprocessMode(SupportedEffectConfig &supportedEffectConfig, MessageParcel &reply, int32_t i, int32_t j)
 {
     reply.WriteString(supportedEffectConfig.postProcessNew.stream[i].streamEffectMode[j].mode);
-    int32_t countDev = supportedEffectConfig.postProcessNew.stream[i].streamEffectMode[j].devicePort.size();
+    uint32_t countDev = supportedEffectConfig.postProcessNew.stream[i].streamEffectMode[j].devicePort.size();
     reply.WriteInt32(countDev);
     if (countDev > 0) {
         for (int32_t k = 0; k < countDev; k++) {
@@ -683,7 +683,7 @@ static void PostprocessProcess(SupportedEffectConfig &supportedEffectConfig, Mes
 {
     // i th stream
     reply.WriteString(supportedEffectConfig.postProcessNew.stream[i].scene);
-    int countMode = supportedEffectConfig.postProcessNew.stream[i].streamEffectMode.size();
+    uint32_t countMode = supportedEffectConfig.postProcessNew.stream[i].streamEffectMode.size();
     reply.WriteInt32(countMode);
     if (countMode > 0) {
         for (int32_t j = 0; j < countMode; j++) {
@@ -699,9 +699,9 @@ void AudioPolicyManagerStub::QueryEffectSceneModeInternal(MessageParcel &data, M
     int32_t ret = QueryEffectSceneMode(supportedEffectConfig); // audio_policy_server.cpp
     CHECK_AND_RETURN_LOG(ret != -1, "default mode is unavailable !");
 
-    int32_t countPre = supportedEffectConfig.preProcessNew.stream.size();
-    int32_t countPost = supportedEffectConfig.postProcessNew.stream.size();
-    int32_t countPostMap = supportedEffectConfig.postProcessSceneMap.size();
+    uint32_t countPre = supportedEffectConfig.preProcessNew.stream.size();
+    uint32_t countPost = supportedEffectConfig.postProcessNew.stream.size();
+    uint32_t countPostMap = supportedEffectConfig.postProcessSceneMap.size();
     reply.WriteInt32(countPre);
     reply.WriteInt32(countPost);
     reply.WriteInt32(countPostMap);
diff --git a/services/audio_policy/server/src/service/audio_policy_service.cpp b/services/audio_policy/server/src/service/audio_policy_service.cpp
index 2aae1a017bc23c54c7d749b4d8defa812dc2fe9a..198f424d01ca2e42582de1ea47fb1260addf00f5 100644
--- a/services/audio_policy/server/src/service/audio_policy_service.cpp
+++ b/services/audio_policy/server/src/service/audio_policy_service.cpp
@@ -5875,7 +5875,8 @@ float AudioPolicyService::GetMaxAmplitude(const int32_t deviceId)
     return 0;
 }
 
-int32_t AudioPolicyService::OpenPortAndInsertIOHandle(const std::string &moduleName, const AudioModuleInfo &moduleInfo)
+AudioIOHandle AudioPolicyService::OpenPortAndInsertIOHandle(const std::string &moduleName,
+    const AudioModuleInfo &moduleInfo)
 {
     AudioIOHandle ioHandle = audioPolicyManager_.OpenAudioPort(moduleInfo);
     CHECK_AND_RETURN_RET_LOG(ioHandle != OPEN_PORT_FAILURE, ERR_INVALID_HANDLE, "OpenAudioPort failed %{public}d",
diff --git a/services/audio_policy/server/src/service/config/audio_device_parser.cpp b/services/audio_policy/server/src/service/config/audio_device_parser.cpp
index 9c8605b06ae9166294699cea1bae87b8af4ab1b9..ea65081c56a5c11040083cd1f891e1d30e232b01 100644
--- a/services/audio_policy/server/src/service/config/audio_device_parser.cpp
+++ b/services/audio_policy/server/src/service/config/audio_device_parser.cpp
@@ -193,7 +193,7 @@ DeviceNodeName AudioDeviceParser::GetDeviceNodeNameAsInt(xmlNode *node)
 std::vector<std::string> split(const std::string &line, const std::string &sep)
 {
     std::vector<std::string> buf;
-    int temp = 0;
+    size_t temp = 0;
     std::string::size_type pos = 0;
     while (true) {
         pos = line.find(sep, temp);
diff --git a/services/audio_policy/server/src/service/config/audio_strategy_router_parser.cpp b/services/audio_policy/server/src/service/config/audio_strategy_router_parser.cpp
index 777eb22ae524dc1ab71c4464dc365bdc0ac6f6cc..878de1f660e0632af04195e3cb6bcc00cb0f2c3b 100644
--- a/services/audio_policy/server/src/service/config/audio_strategy_router_parser.cpp
+++ b/services/audio_policy/server/src/service/config/audio_strategy_router_parser.cpp
@@ -114,7 +114,7 @@ std::string AudioStrategyRouterParser::ExtractPropertyValue(const std::string &p
 std::vector<std::string> AudioStrategyRouterParser::split(const std::string &line, const std::string &sep)
 {
     std::vector<std::string> buf;
-    int temp = 0;
+    size_t temp = 0;
     std::string::size_type pos = 0;
     while (true) {
         pos = line.find(sep, temp);
diff --git a/services/audio_policy/server/src/service/config/audio_usage_strategy_parser.cpp b/services/audio_policy/server/src/service/config/audio_usage_strategy_parser.cpp
index 392333aa45c98ab845fa9bd79f2031b07512f577..3381091fd5fa22c7f0ec1c03b2b1e1f499286921 100644
--- a/services/audio_policy/server/src/service/config/audio_usage_strategy_parser.cpp
+++ b/services/audio_policy/server/src/service/config/audio_usage_strategy_parser.cpp
@@ -108,7 +108,7 @@ void AudioUsageStrategyParser::ParserSourceTypeList(xmlNode *node)
 std::vector<std::string> AudioUsageStrategyParser::split(const std::string &line, const std::string &sep)
 {
     std::vector<std::string> buf;
-    int temp = 0;
+    size_t temp = 0;
     std::string::size_type pos = 0;
     while (true) {
         pos = line.find(sep, temp);
diff --git a/services/audio_policy/server/src/service/manager/audio_adapter_manager.cpp b/services/audio_policy/server/src/service/manager/audio_adapter_manager.cpp
index 72798c0b91b692f780f1962e3677218c5021dac5..a0f4de538ac33dcfba622f2eb04319d9cf7c19c8 100644
--- a/services/audio_policy/server/src/service/manager/audio_adapter_manager.cpp
+++ b/services/audio_policy/server/src/service/manager/audio_adapter_manager.cpp
@@ -1541,7 +1541,7 @@ float AudioAdapterManager::CalculateVolumeDbNonlinear(AudioStreamType streamType
     DeviceVolumeType deviceCategory = GetDeviceCategory(deviceType);
     std::vector<VolumePoint> volumePoints;
     GetVolumePoints(streamAlias, deviceCategory, volumePoints);
-    int32_t pointSize = volumePoints.size();
+    uint32_t pointSize = volumePoints.size();
 
     int32_t volSteps = 1 + volumePoints[pointSize - 1].index - volumePoints[0].index;
     int32_t idxRatio = (volSteps * (volumeLevel - minVolIndex)) / (maxVolIndex - minVolIndex);
diff --git a/services/audio_service/client/src/audio_manager_proxy.cpp b/services/audio_service/client/src/audio_manager_proxy.cpp
index 9d8868f07039050045a86fbb866e3cdf9350a4dd..ec5b1991f601955d1db1b7b0647330e37515d4dc 100644
--- a/services/audio_service/client/src/audio_manager_proxy.cpp
+++ b/services/audio_service/client/src/audio_manager_proxy.cpp
@@ -629,8 +629,8 @@ bool AudioManagerProxy::LoadAudioEffectLibraries(const vector<Library> libraries
     bool ret = dataParcel.WriteInterfaceToken(GetDescriptor());
     CHECK_AND_RETURN_RET_LOG(ret, false, "WriteInterfaceToken failed");
 
-    int32_t countLib = libraries.size();
-    int32_t countEff = effects.size();
+    uint32_t countLib = libraries.size();
+    uint32_t countEff = effects.size();
 
     dataParcel.WriteInt32(countLib);
     dataParcel.WriteInt32(countEff);
@@ -690,7 +690,7 @@ bool AudioManagerProxy::CreateEffectChainManager(std::vector<EffectChain> &effec
     bool ret = dataParcel.WriteInterfaceToken(GetDescriptor());
     CHECK_AND_RETURN_RET_LOG(ret, false, "WriteInterfaceToken failed");
 
-    int32_t countEffectChains = effectChains.size();
+    uint32_t countEffectChains = effectChains.size();
     std::vector<int32_t> listCountEffects;
 
     for (EffectChain &effectChain: effectChains) {
diff --git a/services/audio_service/client/src/audio_process_in_client.cpp b/services/audio_service/client/src/audio_process_in_client.cpp
index 19e6558890eca7852c5fbd79eb18ad48c4af003d..d7871a3d35ccdb4c0bb0944d86c5002fccba2d13 100644
--- a/services/audio_service/client/src/audio_process_in_client.cpp
+++ b/services/audio_service/client/src/audio_process_in_client.cpp
@@ -430,7 +430,7 @@ void AudioProcessInClientInner::SetPreferredFrameSize(int32_t frameSize)
 {
     size_t originalSpanSizeInFrame = static_cast<size_t>(spanSizeInFrame_);
     size_t tmp = static_cast<size_t>(frameSize);
-    size_t count = frameSize / spanSizeInFrame_;
+    size_t count = static_cast<size_t>(frameSize / spanSizeInFrame_);
     size_t rest = frameSize % spanSizeInFrame_;
     if (tmp <= originalSpanSizeInFrame) {
         clientSpanSizeInFrame_ = originalSpanSizeInFrame;
@@ -1536,7 +1536,7 @@ bool AudioProcessInClientInner::PrepareNextIndependent(uint64_t curWritePos, int
     uint64_t nextHandlePos = curWritePos + spanSizeInFrame_;
     Trace prepareTrace("AudioEndpoint::PrepareNextLoop " + std::to_string(nextHandlePos));
     int64_t nextHdiReadTime = GetPredictNextHandleTime(nextHandlePos, true);
-    int64_t aheadTime = spanSizeInFrame_ * AUDIO_NS_PER_SECOND / processConfig_.streamInfo.samplingRate;
+    uint64_t aheadTime = spanSizeInFrame_ * AUDIO_NS_PER_SECOND / processConfig_.streamInfo.samplingRate;
     int64_t nextServerHandleTime = nextHdiReadTime - aheadTime;
     if (nextServerHandleTime < ClockTime::GetCurNano()) {
         wakeUpTime = ClockTime::GetCurNano() + ONE_MILLISECOND_DURATION; // make sure less than duration
diff --git a/services/audio_service/client/src/audio_service_client.cpp b/services/audio_service/client/src/audio_service_client.cpp
index 3c73a374fb428498f9fc98f7f3699f6a307790d7..f8c779b1e5325725f141707d0204975ff2282a85 100644
--- a/services/audio_service/client/src/audio_service_client.cpp
+++ b/services/audio_service/client/src/audio_service_client.cpp
@@ -2336,7 +2336,7 @@ void AudioServiceClient::GetOffloadApproximatelyCacheTime(uint64_t paTimeStamp,
         ppTimeStamp = timeNowSteady;
         int64_t timeDelta = static_cast<int64_t>(timeNowSteady) - static_cast<int64_t>(offloadLastHdiPosTs_);
         timeDelta = timeDelta > 0 ? timeDelta : 0;
-        frames = offloadLastHdiPosFrames_ + timeDelta;
+        frames = offloadLastHdiPosFrames_ + static_cast<uint64_t>(timeDelta);
     }
 
     int64_t timeDelta = static_cast<int64_t>(paTimeStamp) - static_cast<int64_t>(ppTimeStamp);
@@ -2394,7 +2394,7 @@ int32_t AudioServiceClient::GetCurrentPosition(uint64_t &framePosition, uint64_t
 
 void AudioServiceClient::GetAudioLatencyOffload(uint64_t &latency)
 {
-    int64_t timeNow = static_cast<int64_t>(std::chrono::duration_cast<std::chrono::microseconds>(
+    uint64_t timeNow = static_cast<int64_t>(std::chrono::duration_cast<std::chrono::microseconds>(
         std::chrono::system_clock::now().time_since_epoch()).count());
     bool offloadPwrActive = offloadStatePolicy_ != OFFLOAD_INACTIVE_BACKGROUND;
     if (offloadPwrActive ||
diff --git a/services/audio_service/client/src/audio_spatial_channel_converter.cpp b/services/audio_service/client/src/audio_spatial_channel_converter.cpp
index 7fef760c8bea9b3fd9812965dafa6b86d594786b..059d0224fffca44a5297791c62f68a649a830996 100644
--- a/services/audio_service/client/src/audio_spatial_channel_converter.cpp
+++ b/services/audio_service/client/src/audio_spatial_channel_converter.cpp
@@ -236,7 +236,7 @@ bool LibLoader::AddAlgoHandle(Library library)
 bool LibLoader::Init()
 {
     int32_t ret = 0;
-    int32_t replyData = 0;
+    uint32_t replyData = 0;
     AudioEffectTransInfo cmdInfo = {sizeof(AudioEffectConfig), &ioBufferConfig_};
     AudioEffectTransInfo replyInfo = {sizeof(int32_t), &replyData};
     ret = (*handle_)->command(handle_, EFFECT_CMD_INIT, &cmdInfo, &replyInfo);
diff --git a/services/audio_service/client/src/audio_stream_manager.cpp b/services/audio_service/client/src/audio_stream_manager.cpp
index 383fca3e58335ae4d8af5f385c2c0c17204b7917..4fdab5c97ae660f4ce3bf6ca3d9c6ea4e4177b56 100644
--- a/services/audio_service/client/src/audio_stream_manager.cpp
+++ b/services/audio_service/client/src/audio_stream_manager.cpp
@@ -107,7 +107,7 @@ int32_t AudioStreamManager::GetEffectInfoArray(AudioSceneEffectInfo &audioSceneE
     std::string effectScene = IAudioStream::GetEffectSceneName(streamUsage);
     SupportedEffectConfig supportedEffectConfig;
     int32_t ret = AudioPolicyManager::GetInstance().QueryEffectSceneMode(supportedEffectConfig);
-    int32_t streamNum = supportedEffectConfig.postProcessNew.stream.size();
+    uint32_t streamNum = supportedEffectConfig.postProcessNew.stream.size();
     if (streamNum >= 0) {
         int32_t sceneFlag = 0;
         for (int32_t i = 0; i < streamNum; i++) {
diff --git a/services/audio_service/client/src/audio_system_manager.cpp b/services/audio_service/client/src/audio_system_manager.cpp
index 2ea97d71482670304b887c236ce199fde38b55e2..224e1cfc58c34d3591e8397681b64972c1987b4b 100644
--- a/services/audio_service/client/src/audio_system_manager.cpp
+++ b/services/audio_service/client/src/audio_system_manager.cpp
@@ -684,7 +684,7 @@ int32_t AudioSystemManager::RegisterFocusInfoChangeCallback(
 {
     CHECK_AND_RETURN_RET_LOG(callback != nullptr, ERR_INVALID_PARAM, "callback is null");
 
-    int32_t clientId = GetCallingPid();
+    int32_t clientId = static_cast<int32_t>(GetCallingPid());
     AUDIO_DEBUG_LOG("RegisterFocusInfoChangeCallback clientId:%{public}d", clientId);
     if (audioFocusInfoCallback_ == nullptr) {
         audioFocusInfoCallback_ = std::make_shared<AudioFocusInfoChangeCallbackImpl>();
@@ -706,7 +706,7 @@ int32_t AudioSystemManager::RegisterFocusInfoChangeCallback(
 int32_t AudioSystemManager::UnregisterFocusInfoChangeCallback(
     const std::shared_ptr<AudioFocusInfoChangeCallback> &callback)
 {
-    int32_t clientId = GetCallingPid();
+    int32_t clientId = static_cast<int32_t>(GetCallingPid());
     int32_t ret = 0;
 
     if (callback == nullptr) {
@@ -896,7 +896,7 @@ int32_t AudioSystemManager::DeactivateAudioInterrupt(const AudioInterrupt &audio
 
 int32_t AudioSystemManager::SetAudioManagerInterruptCallback(const std::shared_ptr<AudioManagerCallback> &callback)
 {
-    int32_t clientId = GetCallingPid();
+    int32_t clientId = static_cast<int32_t>(GetCallingPid());
     AUDIO_INFO_LOG("client id: %{public}d", clientId);
     CHECK_AND_RETURN_RET_LOG(callback != nullptr, ERR_INVALID_PARAM, "callback is null");
 
@@ -926,7 +926,7 @@ int32_t AudioSystemManager::SetAudioManagerInterruptCallback(const std::shared_p
 
 int32_t AudioSystemManager::UnsetAudioManagerInterruptCallback()
 {
-    int32_t clientId = GetCallingPid();
+    int32_t clientId = static_cast<int32_t>(GetCallingPid());
     AUDIO_INFO_LOG("client id: %{public}d", clientId);
 
     int32_t ret = AudioPolicyManager::GetInstance().UnsetAudioManagerInterruptCallback(clientId);
@@ -940,7 +940,7 @@ int32_t AudioSystemManager::UnsetAudioManagerInterruptCallback()
 
 int32_t AudioSystemManager::RequestAudioFocus(const AudioInterrupt &audioInterrupt)
 {
-    int32_t clientId = GetCallingPid();
+    int32_t clientId = static_cast<int32_t>(GetCallingPid());
     AUDIO_INFO_LOG("RequestAudioFocus client id: %{public}d", clientId);
     CHECK_AND_RETURN_RET_LOG(audioInterrupt.contentType >= CONTENT_TYPE_UNKNOWN &&
         audioInterrupt.contentType <= CONTENT_TYPE_ULTRASONIC, ERR_INVALID_PARAM, "Invalid content type");
@@ -953,7 +953,7 @@ int32_t AudioSystemManager::RequestAudioFocus(const AudioInterrupt &audioInterru
 
 int32_t AudioSystemManager::AbandonAudioFocus(const AudioInterrupt &audioInterrupt)
 {
-    int32_t clientId = GetCallingPid();
+    int32_t clientId = static_cast<int32_t>(GetCallingPid());
     AUDIO_INFO_LOG("AbandonAudioFocus client id: %{public}d", clientId);
     CHECK_AND_RETURN_RET_LOG(audioInterrupt.contentType >= CONTENT_TYPE_UNKNOWN &&
         audioInterrupt.contentType <= CONTENT_TYPE_ULTRASONIC, ERR_INVALID_PARAM, "Invalid content type");
@@ -1039,7 +1039,7 @@ bool AudioSystemManager::RequestIndependentInterrupt(FocusType focusType)
 {
     AUDIO_INFO_LOG("RequestIndependentInterrupt : foncusType");
     AudioInterrupt audioInterrupt;
-    int32_t clientId = GetCallingPid();
+    int32_t clientId = static_cast<int32_t>(GetCallingPid());
     audioInterrupt.contentType = ContentType::CONTENT_TYPE_SPEECH;
     audioInterrupt.streamUsage = StreamUsage::STREAM_USAGE_MEDIA;
     audioInterrupt.audioFocusType.streamType = AudioStreamType::STREAM_RECORDING;
@@ -1053,7 +1053,7 @@ bool AudioSystemManager::AbandonIndependentInterrupt(FocusType focusType)
 {
     AUDIO_INFO_LOG("AbandonIndependentInterrupt : foncusType");
     AudioInterrupt audioInterrupt;
-    int32_t clientId = GetCallingPid();
+    int32_t clientId = static_cast<int32_t>(GetCallingPid());
     audioInterrupt.contentType = ContentType::CONTENT_TYPE_SPEECH;
     audioInterrupt.streamUsage = StreamUsage::STREAM_USAGE_MEDIA;
     audioInterrupt.audioFocusType.streamType = AudioStreamType::STREAM_RECORDING;
diff --git a/services/audio_service/client/src/capturer_in_client.cpp b/services/audio_service/client/src/capturer_in_client.cpp
index 7d947935b3ef8986c626effa238d76149fb7666a..a6e36e8336bc17a2293c24b10d3c0570bb4de530 100644
--- a/services/audio_service/client/src/capturer_in_client.cpp
+++ b/services/audio_service/client/src/capturer_in_client.cpp
@@ -290,7 +290,7 @@ private:
     std::mutex readMutex_; // used for prevent multi thread call read
 
     // Mark reach and period reach callback
-    int64_t totalBytesRead_ = 0;
+    uint64_t totalBytesRead_ = 0;
     std::mutex markReachMutex_;
     bool capturerMarkReached_ = false;
     int64_t capturerMarkPosition_ = 0;
@@ -792,7 +792,7 @@ bool CapturerInClientInner::GetAudioTime(Timestamp &timestamp, Timestamp::Timest
 {
     CHECK_AND_RETURN_RET_LOG(paramsIsSet_ == true, false, "Params is not set");
     CHECK_AND_RETURN_RET_LOG(state_ != STOPPED, false, "Invalid status:%{public}d", state_.load());
-    int64_t currentReadPos = totalBytesRead_ / sizePerFrameInByte_;
+    uint64_t currentReadPos = totalBytesRead_ / sizePerFrameInByte_;
     timestamp.framePosition = currentReadPos;
 
     uint64_t writePos = 0;
@@ -1191,7 +1191,7 @@ int64_t CapturerInClientInner::GetFramesWritten()
 int64_t CapturerInClientInner::GetFramesRead()
 {
     CHECK_AND_RETURN_RET_LOG(sizePerFrameInByte_ != 0, INVALID_FRAME_SIZE, "sizePerFrameInByte_ is 0!");
-    int64_t readFrameNumber = totalBytesRead_ / sizePerFrameInByte_;
+    uint64_t readFrameNumber = totalBytesRead_ / sizePerFrameInByte_;
     return readFrameNumber;
 }
 
@@ -1587,7 +1587,7 @@ void CapturerInClientInner::HandleCapturerPositionChanges(size_t bytesRead)
         AUDIO_ERR_LOG("HandleCapturerPositionChanges: sizePerFrameInByte_ is 0");
         return;
     }
-    int64_t readFrameNumber = totalBytesRead_ / sizePerFrameInByte_;
+    uint64_t readFrameNumber = totalBytesRead_ / sizePerFrameInByte_;
     AUDIO_DEBUG_LOG("totalBytesRead_ %{public}" PRId64 ", frame size: %{public}zu", totalBytesRead_,
         sizePerFrameInByte_);
     {
diff --git a/services/audio_service/client/src/renderer_in_client.cpp b/services/audio_service/client/src/renderer_in_client.cpp
index cc75d5c5dcd5a69fd7a46e64a9e01c7c61a761e9..d8ee0ea59e77ac0c93f9be7b08f3a50848c5620b 100644
--- a/services/audio_service/client/src/renderer_in_client.cpp
+++ b/services/audio_service/client/src/renderer_in_client.cpp
@@ -64,7 +64,7 @@ namespace AudioStandard {
 namespace {
 const uint64_t OLD_BUF_DURATION_IN_USEC = 92880; // This value is used for compatibility purposes.
 const uint64_t AUDIO_US_PER_MS = 1000;
-const int64_t AUDIO_NS_PER_US = 1000;
+const uint64_t AUDIO_NS_PER_US = 1000;
 const uint64_t AUDIO_US_PER_S = 1000000;
 const uint64_t AUDIO_MS_PER_S = 1000;
 const uint64_t MAX_BUF_DURATION_IN_USEC = 2000000; // 2S
@@ -321,7 +321,7 @@ private:
     size_t clientSpanSizeInByte_ = 0;
     size_t sizePerFrameInByte_ = 4; // 16bit 2ch as default
 
-    int32_t bufferSizeInMsec_ = 20; // 20ms
+    uint32_t bufferSizeInMsec_ = 20; // 20ms
     std::string cachePath_;
     std::string dumpOutFile_;
     FILE *dumpOutFd_ = nullptr;
@@ -909,7 +909,7 @@ bool RendererInClientInner::GetAudioTime(Timestamp &timestamp, Timestamp::Timest
         int64_t timeNow = static_cast<int64_t>(std::chrono::duration_cast<std::chrono::microseconds>(
             std::chrono::system_clock::now().time_since_epoch()).count());
         int64_t deltaTimeStamp = (static_cast<int64_t>(timeNow) - static_cast<int64_t>(timestampHdi)) * AUDIO_NS_PER_US;
-        int64_t paWriteIndexNs = paWriteIndex * AUDIO_NS_PER_US;
+        uint64_t paWriteIndexNs = paWriteIndex * AUDIO_NS_PER_US;
         uint64_t readPosNs = readPos * AUDIO_MS_PER_SECOND / curStreamParams_.samplingRate * AUDIO_US_PER_S;
 
         int64_t deltaPaWriteIndexNs = static_cast<int64_t>(readPosNs) - static_cast<int64_t>(paWriteIndexNs);
diff --git a/services/audio_service/common/src/linear_pos_time_model.cpp b/services/audio_service/common/src/linear_pos_time_model.cpp
index 751328b0d65f687c6d76b1f7b806ae91a981a72a..919baf4a48fd154e08203f8baea6d7fbf9b16aaf 100644
--- a/services/audio_service/common/src/linear_pos_time_model.cpp
+++ b/services/audio_service/common/src/linear_pos_time_model.cpp
@@ -110,7 +110,7 @@ void LinearPosTimeModel::SetSpanCount(uint64_t spanCountInFrame)
 
 int64_t LinearPosTimeModel::GetTimeOfPos(uint64_t posInFrame)
 {
-    int64_t deltaFrame = 0;
+    uint64_t deltaFrame = 0;
     int64_t invalidTime = -1;
     CHECK_AND_RETURN_RET_LOG(isConfiged, invalidTime, "SampleRate is not configed!");
     if (posInFrame >= stampFrame_) {
diff --git a/services/audio_service/server/src/audio_endpoint.cpp b/services/audio_service/server/src/audio_endpoint.cpp
index 31b2ebc12bb9e89e41fbe0b208851e165157cdf4..83d976cfba4b31abdafd77ad8adceca7bcfbb46c 100644
--- a/services/audio_service/server/src/audio_endpoint.cpp
+++ b/services/audio_service/server/src/audio_endpoint.cpp
@@ -1526,7 +1526,7 @@ void AudioEndpointInner::CheckPlaySignal(uint8_t *buffer, size_t bufferSize)
         return;
     }
     CHECK_AND_RETURN_LOG(signalDetectAgent_ != nullptr, "LatencyMeas signalDetectAgent_ is nullptr");
-    int32_t byteSize = GetFormatByteSize(dstStreamInfo_.format);
+    size_t byteSize = GetFormatByteSize(dstStreamInfo_.format);
     size_t newlyCheckedTime = bufferSize / (dstStreamInfo_.samplingRate /
         MILLISECOND_PER_SECOND) / (byteSize * sizeof(uint8_t) * dstStreamInfo_.channels);
     detectedTime_ += newlyCheckedTime;
diff --git a/services/audio_service/server/src/audio_server.cpp b/services/audio_service/server/src/audio_server.cpp
index 6d5f2f89523ac371a0c63865262d5acf4a2f4a82..5f8a286b6d03895bbf500429d35a8668006cbcfa 100644
--- a/services/audio_service/server/src/audio_server.cpp
+++ b/services/audio_service/server/src/audio_server.cpp
@@ -1257,7 +1257,7 @@ void AudioServer::RequestThreadPriority(uint32_t tid, string bundleName)
 {
     AUDIO_INFO_LOG("RequestThreadPriority tid: %{public}u", tid);
 
-    uint32_t pid = IPCSkeleton::GetCallingPid();
+    int32_t pid = IPCSkeleton::GetCallingPid();
     AudioXCollie audioXCollie("AudioServer::ScheduleReportData", SCHEDULE_REPORT_TIME_OUT_SECONDS);
     ScheduleReportData(pid, tid, bundleName.c_str());
 }
diff --git a/services/audio_service/server/src/pa_renderer_stream_impl.cpp b/services/audio_service/server/src/pa_renderer_stream_impl.cpp
index 369da6157dda27880b8a6fb74d133903c207766a..1f139624bac09253e2f6a6c280042169c51bbc79 100644
--- a/services/audio_service/server/src/pa_renderer_stream_impl.cpp
+++ b/services/audio_service/server/src/pa_renderer_stream_impl.cpp
@@ -863,7 +863,7 @@ int32_t PaRendererStreamImpl::GetOffloadApproximatelyCacheTime(uint64_t &timesta
     timestamp = info->timestamp.tv_sec * AUDIO_US_PER_SECOND + info->timestamp.tv_usec;
     lock.Unlock();
 
-    int64_t cacheTimeInPulse = writeIndex > readIndex ? writeIndex - readIndex : 0;
+    uint64_t cacheTimeInPulse = writeIndex > readIndex ? writeIndex - readIndex : 0;
     cacheTimePa = static_cast<uint64_t>(cacheTimeInPulse);
     paWriteIndex = writeIndex;