| /* | |
| * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | |
| * | |
| * Use of this source code is governed by a BSD-style license | |
| * that can be found in the LICENSE file in the root of the source | |
| * tree. An additional intellectual property rights grant can be found | |
| * in the file PATENTS. All contributing project authors may | |
| * be found in the AUTHORS file in the root of the source tree. | |
| */ | |
| #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_PROTOBUF_UTILS_H_ | |
| #define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_PROTOBUF_UTILS_H_ | |
| #include "webrtc/audio_processing/debug.pb.h" | |
| #include "webrtc/base/scoped_ptr.h" | |
| namespace webrtc { | |
| // Allocates new memory in the scoped_ptr to fit the raw message and returns the | |
| // number of bytes read. | |
| 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); | |
| } // namespace webrtc | |
| #endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_PROTOBUF_UTILS_H_ |