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/*
* 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_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_AUDIO_NETWORK_ADAPTOR_H_
#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_AUDIO_NETWORK_ADAPTOR_H_
#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "webrtc/test/gmock.h"
namespace webrtc {
class MockAudioNetworkAdaptor : public AudioNetworkAdaptor {
public:
virtual ~MockAudioNetworkAdaptor() { Die(); }
MOCK_METHOD0(Die, void());
MOCK_METHOD1(SetUplinkBandwidth, void(int uplink_bandwidth_bps));
MOCK_METHOD1(SetUplinkPacketLossFraction,
void(float uplink_packet_loss_fraction));
MOCK_METHOD1(SetUplinkRecoverablePacketLossFraction,
void(float uplink_recoverable_packet_loss_fraction));
MOCK_METHOD1(SetRtt, void(int rtt_ms));
MOCK_METHOD1(SetTargetAudioBitrate, void(int target_audio_bitrate_bps));
MOCK_METHOD1(SetOverhead, void(size_t overhead_bytes_per_packet));
MOCK_METHOD0(GetEncoderRuntimeConfig, AudioEncoderRuntimeConfig());
MOCK_METHOD1(StartDebugDump, void(FILE* file_handle));
MOCK_METHOD0(StopDebugDump, void());
MOCK_CONST_METHOD0(GetStats, ANAStats());
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_AUDIO_NETWORK_ADAPTOR_H_