Index: webrtc/call/mock/mock_rtc_event_log.h |
diff --git a/webrtc/call/mock/mock_rtc_event_log.h b/webrtc/call/mock/mock_rtc_event_log.h |
new file mode 100644 |
index 0000000000000000000000000000000000000000..f523105d0e40d666c033b0f24b8dc4953f608b17 |
--- /dev/null |
+++ b/webrtc/call/mock/mock_rtc_event_log.h |
@@ -0,0 +1,61 @@ |
+/* |
+ * 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 "testing/gmock/include/gmock/gmock.h" |
+ |
+#include "webrtc/call/rtc_event_log.h" |
+ |
+namespace webrtc { |
+ |
+class MockRtcEventLog : public RtcEventLog { |
+ public: |
+ MOCK_METHOD1(SetBufferDuration, void(int64_t buffer_duration_us)); |
+ |
+ MOCK_METHOD2(StartLogging, |
+ void(const std::string& file_name, int duration_ms)); |
+ |
+ MOCK_METHOD1(StartLogging, bool(rtc::PlatformFile log_file)); |
+ |
+ 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_ |