ac9f876bc0
gmock.h and gtest.h were moved (or rather, got wrappers so that we could put some icky compatibility hacks in one place instead of 500) in this CL: https://codereview.webrtc.org/2358993004/ NOPRESUBMIT=true BUG=webrtc:6398 Review-Url: https://codereview.webrtc.org/2381013002 Cr-Commit-Position: refs/heads/master@{#14464}
60 lines
1.8 KiB
C++
60 lines
1.8 KiB
C++
/*
|
|
* Copyright (c) 2016 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_CALL_MOCK_MOCK_RTC_EVENT_LOG_H_
|
|
#define WEBRTC_CALL_MOCK_MOCK_RTC_EVENT_LOG_H_
|
|
|
|
#include <string>
|
|
|
|
#include "webrtc/call/rtc_event_log.h"
|
|
#include "webrtc/test/gmock.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class MockRtcEventLog : public RtcEventLog {
|
|
public:
|
|
MOCK_METHOD2(StartLogging,
|
|
bool(const std::string& file_name, int64_t max_size_bytes));
|
|
|
|
MOCK_METHOD2(StartLogging,
|
|
bool(rtc::PlatformFile log_file, int64_t max_size_bytes));
|
|
|
|
MOCK_METHOD0(StopLogging, void());
|
|
|
|
MOCK_METHOD1(LogVideoReceiveStreamConfig,
|
|
void(const webrtc::VideoReceiveStream::Config& config));
|
|
|
|
MOCK_METHOD1(LogVideoSendStreamConfig,
|
|
void(const webrtc::VideoSendStream::Config& config));
|
|
|
|
MOCK_METHOD4(LogRtpHeader,
|
|
void(PacketDirection direction,
|
|
MediaType media_type,
|
|
const uint8_t* header,
|
|
size_t packet_length));
|
|
|
|
MOCK_METHOD4(LogRtcpPacket,
|
|
void(PacketDirection direction,
|
|
MediaType media_type,
|
|
const uint8_t* packet,
|
|
size_t length));
|
|
|
|
MOCK_METHOD1(LogAudioPlayout, void(uint32_t ssrc));
|
|
|
|
MOCK_METHOD3(LogBwePacketLossEvent,
|
|
void(int32_t bitrate,
|
|
uint8_t fraction_loss,
|
|
int32_t total_packets));
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_CALL_MOCK_MOCK_RTC_EVENT_LOG_H_
|