| /* |
| * 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_ |