Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/neteq/

webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc

Index: webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc b/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
index faabdc241c80bdb396dad3922138f50edb5b4c74..0735b4c388d50b8672a99a9e49f3025188409e71 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
@@ -10,10 +10,11 @@
 
 #include <assert.h>
 #include <stdio.h>
+
+#include <memory>
 #include <vector>
 
 #include "gflags/gflags.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
 
@@ -63,7 +64,7 @@ int main(int argc, char* argv[]) {
   }
 
   printf("Input file: %s\n", argv[1]);
-  rtc::scoped_ptr<webrtc::test::RtpFileSource> file_source(
+  std::unique_ptr<webrtc::test::RtpFileSource> file_source(
       webrtc::test::RtpFileSource::Create(argv[1]));
   assert(file_source.get());
   // Set RTP extension IDs.
@@ -104,7 +105,7 @@ int main(int argc, char* argv[]) {
 
   uint32_t max_abs_send_time = 0;
   int cycles = -1;
-  rtc::scoped_ptr<webrtc::test::Packet> packet;
+  std::unique_ptr<webrtc::test::Packet> packet;
   while (true) {
     packet.reset(file_source->NextPacket());
     if (!packet.get()) {