Index: webrtc/modules/audio_processing/test/protobuf_utils.cc |
diff --git a/webrtc/modules/audio_processing/test/protobuf_utils.cc b/webrtc/modules/audio_processing/test/protobuf_utils.cc |
index 37042cdc141a5de8b99f2aff937b7966a6018397..c18a13e6edfe306688491efe17f834169484e9d3 100644 |
--- a/webrtc/modules/audio_processing/test/protobuf_utils.cc |
+++ b/webrtc/modules/audio_processing/test/protobuf_utils.cc |
@@ -9,10 +9,11 @@ |
*/ |
#include "webrtc/modules/audio_processing/test/protobuf_utils.h" |
+#include "webrtc/typedefs.h" |
namespace webrtc { |
-size_t ReadMessageBytesFromFile(FILE* file, rtc::scoped_ptr<uint8_t[]>* bytes) { |
+size_t ReadMessageBytesFromFile(FILE* file, std::unique_ptr<uint8_t[]>* bytes) { |
// The "wire format" for the size is little-endian. Assume we're running on |
// a little-endian machine. |
#ifndef WEBRTC_ARCH_LITTLE_ENDIAN |
@@ -30,7 +31,7 @@ size_t ReadMessageBytesFromFile(FILE* file, rtc::scoped_ptr<uint8_t[]>* bytes) { |
// Returns true on success, false on error or end-of-file. |
bool ReadMessageFromFile(FILE* file, ::google::protobuf::MessageLite* msg) { |
- rtc::scoped_ptr<uint8_t[]> bytes; |
+ std::unique_ptr<uint8_t[]> bytes; |
size_t size = ReadMessageBytesFromFile(file, &bytes); |
if (!size) |
return false; |