| Index: webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc | 
| diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc | 
| index b7a3109c01f0b33223f3bb46876d92f10910bde5..039e1fae2e57a9fd8f07ca96a13d234d14f81a90 100644 | 
| --- a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc | 
| +++ b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc | 
| @@ -18,6 +18,8 @@ | 
| #include <netinet/in.h> | 
| #endif | 
|  | 
| +#include <memory> | 
| + | 
| #include "webrtc/base/checks.h" | 
| #include "webrtc/modules/audio_coding/neteq/tools/packet.h" | 
| #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" | 
| @@ -33,13 +35,13 @@ RtpFileSource* RtpFileSource::Create(const std::string& file_name) { | 
| } | 
|  | 
| bool RtpFileSource::ValidRtpDump(const std::string& file_name) { | 
| -  rtc::scoped_ptr<RtpFileReader> temp_file( | 
| +  std::unique_ptr<RtpFileReader> temp_file( | 
| RtpFileReader::Create(RtpFileReader::kRtpDump, file_name)); | 
| return !!temp_file; | 
| } | 
|  | 
| bool RtpFileSource::ValidPcap(const std::string& file_name) { | 
| -  rtc::scoped_ptr<RtpFileReader> temp_file( | 
| +  std::unique_ptr<RtpFileReader> temp_file( | 
| RtpFileReader::Create(RtpFileReader::kPcap, file_name)); | 
| return !!temp_file; | 
| } | 
| @@ -64,9 +66,9 @@ Packet* RtpFileSource::NextPacket() { | 
| // Read the next one. | 
| continue; | 
| } | 
| -    rtc::scoped_ptr<uint8_t[]> packet_memory(new uint8_t[temp_packet.length]); | 
| +    std::unique_ptr<uint8_t[]> packet_memory(new uint8_t[temp_packet.length]); | 
| memcpy(packet_memory.get(), temp_packet.data, temp_packet.length); | 
| -    rtc::scoped_ptr<Packet> packet(new Packet( | 
| +    std::unique_ptr<Packet> packet(new Packet( | 
| packet_memory.release(), temp_packet.length, | 
| temp_packet.original_length, temp_packet.time_ms, *parser_.get())); | 
| if (!packet->valid_header()) { | 
|  |