diff --git a/src/channel/rtp_channel/rtp_video_receiver.h b/src/channel/rtp_channel/rtp_video_receiver.h
index b708516..0f6d92b 100644
--- a/src/channel/rtp_channel/rtp_video_receiver.h
+++ b/src/channel/rtp_channel/rtp_video_receiver.h
@@ -14,7 +14,7 @@
 #include "rtcp_receiver_report.h"
 #include "rtp_packet_av1.h"
 #include "rtp_packet_h264.h"
-#include "rtp_rtcp_defines.h "
+#include "rtp_rtcp_defines.h"
 #include "rtp_statistics.h"
 #include "thread_base.h"
 #include "video_frame.h"
diff --git a/src/common/copy_on_write_buffer.h b/src/common/copy_on_write_buffer.h
index 16aacfb..eaf70c4 100644
--- a/src/common/copy_on_write_buffer.h
+++ b/src/common/copy_on_write_buffer.h
@@ -38,7 +38,7 @@ class CopyOnWriteBuffer {
 
  private:
   void EnsureUnique() {
-    if (!buffer_.unique()) {
+    if (buffer_.use_count() != 1) {
       buffer_ = std::make_shared<std::vector<uint8_t>>(*buffer_);
     }
   }
diff --git a/src/common/rtc_base/ref_counter.h b/src/common/rtc_base/ref_counter.h
index 7983c89..af25391 100644
--- a/src/common/rtc_base/ref_counter.h
+++ b/src/common/rtc_base/ref_counter.h
@@ -12,7 +12,7 @@
 
 #include <atomic>
 
-#include "ref_count.h"
+#include "api/ref_count.h"
 
 namespace webrtc {
 namespace webrtc_impl {
diff --git a/src/common/rtc_base/system_time.cc b/src/common/rtc_base/system_time.cc
index 6ae171b..aae087b 100644
--- a/src/common/rtc_base/system_time.cc
+++ b/src/common/rtc_base/system_time.cc
@@ -18,10 +18,10 @@
 
 #if defined(__POSIX__)
 #include <sys/time.h>
+#endif
 #if defined(__APPLE__)
 #include <mach/mach_time.h>
 #endif
-#endif
 
 #if defined(_WIN32)
 // clang-format off
diff --git a/src/common/rtc_base/time_utils.cc b/src/common/rtc_base/time_utils.cc
index b341da9..daf983c 100644
--- a/src/common/rtc_base/time_utils.cc
+++ b/src/common/rtc_base/time_utils.cc
@@ -15,13 +15,12 @@
 #endif
 
 #include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/system_time.h"
 #include "rtc_base/time_utils.h"
-#include "system_time.h"
-#if defined(_WIN32)
-#include "rtc_base/win32.h"
-#endif
 #if defined(_WIN32)
 #include <minwinbase.h>
+
+#include "rtc_base/win32.h"
 #endif
 
 namespace rtc {
@@ -228,6 +227,7 @@ int64_t TimeUTCMicros() {
   return (li.QuadPart - kFileTimeToUnixTimeEpochOffset) /
          kFileTimeToMicroSeconds;
 #endif
+  return 0;
 }
 
 int64_t TimeUTCMillis() { return TimeUTCMicros() / kNumMicrosecsPerMillisec; }
diff --git a/src/common/rtc_base/time_utils.h b/src/common/rtc_base/time_utils.h
index 7c80cb2..98bbc59 100644
--- a/src/common/rtc_base/time_utils.h
+++ b/src/common/rtc_base/time_utils.h
@@ -14,7 +14,7 @@
 #include <stdint.h>
 #include <time.h>
 
-#include "system_time.h"
+#include "rtc_base/system_time.h"
 
 namespace rtc {
 
diff --git a/src/qos/transport_feedback_adapter.cc b/src/qos/transport_feedback_adapter.cc
index 243d509..a313660 100644
--- a/src/qos/transport_feedback_adapter.cc
+++ b/src/qos/transport_feedback_adapter.cc
@@ -96,6 +96,7 @@ void TransportFeedbackAdapter::AddPacket(const RtpPacketToSend& packet_to_send,
   feedback.sent.sequence_number = seq_num_unwrapper_.Unwrap(
       packet_to_send.transport_sequence_number().value_or(0));
   feedback.sent.size = DataSize::Bytes(packet_to_send.size() + overhead_bytes);
+  LOG_WARN("!!! packet_to_send.size: {}", packet_to_send.size());
   feedback.sent.audio =
       packet_to_send.packet_type() == RtpPacketMediaType::kAudio;
   feedback.network_route = network_route_;
diff --git a/src/rtp/rtp_packetizer/rtp_packetizer_generic.cpp b/src/rtp/rtp_packetizer/rtp_packetizer_generic.cpp
index 7d5ff2d..a412bd0 100644
--- a/src/rtp/rtp_packetizer/rtp_packetizer_generic.cpp
+++ b/src/rtp/rtp_packetizer/rtp_packetizer_generic.cpp
@@ -24,9 +24,9 @@ std::vector<RtpPacket> RtpPacketizerGeneric::Build(uint8_t* payload,
       payload_size / MAX_NALU_LEN + (last_packet_size ? 1 : 0);
 
   // TODO: use frame timestamp
-  timestamp_ = std::chrono::duration_cast<std::chrono::microseconds>(
-                   std::chrono::system_clock::now().time_since_epoch())
-                   .count();
+  uint64_t timestamp = std::chrono::duration_cast<std::chrono::microseconds>(
+                           std::chrono::system_clock::now().time_since_epoch())
+                           .count();
 
   std::vector<RtpPacket> rtp_packets;
   for (uint32_t index = 0; index < packet_num; index++) {
@@ -37,11 +37,10 @@ std::vector<RtpPacket> RtpPacketizerGeneric::Build(uint8_t* payload,
     marker_ = index == packet_num - 1 ? 1 : 0;
     payload_type_ = rtp::PAYLOAD_TYPE(payload_type_);
     sequence_number_++;
-    timestamp_ = timestamp_;
+    timestamp_ = timestamp;
     ssrc_ = ssrc_;
 
     if (!csrc_count_) {
-      csrcs_ = csrcs_;
     }
 
     rtp_packet_frame_.clear();
diff --git a/src/rtp/rtp_packetizer/rtp_packetizer_h264.cpp b/src/rtp/rtp_packetizer/rtp_packetizer_h264.cpp
index 408123d..54b11e9 100644
--- a/src/rtp/rtp_packetizer/rtp_packetizer_h264.cpp
+++ b/src/rtp/rtp_packetizer/rtp_packetizer_h264.cpp
@@ -81,10 +81,8 @@ std::vector<RtpPacket> RtpPacketizerH264::BuildNalu(uint8_t* payload,
   timestamp_ = std::chrono::duration_cast<std::chrono::microseconds>(
                    std::chrono::system_clock::now().time_since_epoch())
                    .count();
-  ssrc_ = ssrc_;
 
   if (!csrc_count_) {
-    csrcs_ = csrcs_;
   }
 
   rtp::FU_INDICATOR fu_indicator;
@@ -141,9 +139,9 @@ std::vector<RtpPacket> RtpPacketizerH264::BuildFua(uint8_t* payload,
       payload_size / MAX_NALU_LEN + (last_packet_size ? 1 : 0);
 
   // TODO: use frame timestamp
-  timestamp_ = std::chrono::duration_cast<std::chrono::microseconds>(
-                   std::chrono::system_clock::now().time_since_epoch())
-                   .count();
+  uint64_t timestamp = std::chrono::duration_cast<std::chrono::microseconds>(
+                           std::chrono::system_clock::now().time_since_epoch())
+                           .count();
 
   for (uint32_t index = 0; index < packet_num; index++) {
     version_ = kRtpVersion;
@@ -153,7 +151,7 @@ std::vector<RtpPacket> RtpPacketizerH264::BuildFua(uint8_t* payload,
     marker_ = (index == (packet_num - 1)) ? 1 : 0;
     payload_type_ = rtp::PAYLOAD_TYPE(payload_type_);
     sequence_number_++;
-    timestamp_ = timestamp_;
+    timestamp_ = timestamp;
     ssrc_ = ssrc_;
 
     if (!csrc_count_) {
diff --git a/src/rtp/rtp_packetizer/rtp_packetizer_h264.h b/src/rtp/rtp_packetizer/rtp_packetizer_h264.h
index 05441a1..5c97153 100644
--- a/src/rtp/rtp_packetizer/rtp_packetizer_h264.h
+++ b/src/rtp/rtp_packetizer/rtp_packetizer_h264.h
@@ -18,11 +18,9 @@ class RtpPacketizerH264 : public RtpPacketizer {
   std::vector<RtpPacket> Build(uint8_t* payload,
                                uint32_t payload_size) override;
 
-  std::vector<RtpPacket> RtpPacketizerH264::BuildNalu(uint8_t* payload,
-                                                      uint32_t payload_size);
+  std::vector<RtpPacket> BuildNalu(uint8_t* payload, uint32_t payload_size);
 
-  std::vector<RtpPacket> RtpPacketizerH264::BuildFua(uint8_t* payload,
-                                                     uint32_t payload_size);
+  std::vector<RtpPacket> BuildFua(uint8_t* payload, uint32_t payload_size);
 
  private:
   bool EncodeH264Fua(RtpPacket& rtp_packet, uint8_t* payload,
diff --git a/xmake.lua b/xmake.lua
index 3b2dbbd..4211b1a 100644
--- a/xmake.lua
+++ b/xmake.lua
@@ -48,6 +48,9 @@ target("common")
     "src/common/rtc_base/numerics/*.cc",
     "src/common/api/units/*.cc",
     "src/common/api/transport/*.cc")
+    if not is_os("windows") then
+        remove_files("src/common/rtc_base/win32.cc")
+    end
     add_includedirs("src/common", {public = true})
 
 target("inih")