| 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()) {
|
|
|