blob: 3ab90b7761ac6dab4d47f08d6f852d81b397b440 [file] [log] [blame]
/*
* 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 <memory>
#include "webrtc/modules/audio_processing/debug.pb.h"
namespace webrtc {
// Allocates new memory in the unique_ptr to fit the raw message and returns the
// number of bytes read.
size_t ReadMessageBytesFromFile(FILE* file, std::unique_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_