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()) { |