File: mock_voip_engine.h

package info (click to toggle)
chromium 139.0.7258.127-1
  • links: PTS, VCS
  • area: main
  • in suites:
  • size: 6,122,068 kB
  • sloc: cpp: 35,100,771; ansic: 7,163,530; javascript: 4,103,002; python: 1,436,920; asm: 946,517; xml: 746,709; pascal: 187,653; perl: 88,691; sh: 88,436; objc: 79,953; sql: 51,488; cs: 44,583; fortran: 24,137; makefile: 22,147; tcl: 15,277; php: 13,980; yacc: 8,984; ruby: 7,485; awk: 3,720; lisp: 3,096; lex: 1,327; ada: 727; jsp: 228; sed: 36
file content (126 lines) | stat: -rw-r--r-- 3,887 bytes parent folder | download | duplicates (3)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
/*
 *  Copyright (c) 2021 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 API_VOIP_TEST_MOCK_VOIP_ENGINE_H_
#define API_VOIP_TEST_MOCK_VOIP_ENGINE_H_

#include <cstdint>
#include <map>
#include <optional>

#include "api/array_view.h"
#include "api/audio_codecs/audio_format.h"
#include "api/voip/voip_base.h"
#include "api/voip/voip_codec.h"
#include "api/voip/voip_dtmf.h"
#include "api/voip/voip_engine.h"
#include "api/voip/voip_network.h"
#include "api/voip/voip_statistics.h"
#include "api/voip/voip_volume_control.h"
#include "test/gmock.h"

namespace webrtc {

class MockVoipBase : public VoipBase {
 public:
  MOCK_METHOD(ChannelId,
              CreateChannel,
              (Transport*, std::optional<uint32_t>),
              (override));
  MOCK_METHOD(VoipResult, ReleaseChannel, (ChannelId), (override));
  MOCK_METHOD(VoipResult, StartSend, (ChannelId), (override));
  MOCK_METHOD(VoipResult, StopSend, (ChannelId), (override));
  MOCK_METHOD(VoipResult, StartPlayout, (ChannelId), (override));
  MOCK_METHOD(VoipResult, StopPlayout, (ChannelId), (override));
};

class MockVoipCodec : public VoipCodec {
 public:
  MOCK_METHOD(VoipResult,
              SetSendCodec,
              (ChannelId, int, const SdpAudioFormat&),
              (override));
  MOCK_METHOD(VoipResult,
              SetReceiveCodecs,
              (ChannelId, (const std::map<int, SdpAudioFormat>&)),
              (override));
};

class MockVoipDtmf : public VoipDtmf {
 public:
  MOCK_METHOD(VoipResult,
              RegisterTelephoneEventType,
              (ChannelId, int, int),
              (override));
  MOCK_METHOD(VoipResult,
              SendDtmfEvent,
              (ChannelId, DtmfEvent, int),
              (override));
};

class MockVoipNetwork : public VoipNetwork {
 public:
  MOCK_METHOD(VoipResult,
              ReceivedRTPPacket,
              (ChannelId channel_id, ArrayView<const uint8_t> rtp_packet),
              (override));
  MOCK_METHOD(VoipResult,
              ReceivedRTCPPacket,
              (ChannelId channel_id, ArrayView<const uint8_t> rtcp_packet),
              (override));
};

class MockVoipStatistics : public VoipStatistics {
 public:
  MOCK_METHOD(VoipResult,
              GetIngressStatistics,
              (ChannelId, IngressStatistics&),
              (override));
  MOCK_METHOD(VoipResult,
              GetChannelStatistics,
              (ChannelId channel_id, ChannelStatistics&),
              (override));
};

class MockVoipVolumeControl : public VoipVolumeControl {
 public:
  MOCK_METHOD(VoipResult, SetInputMuted, (ChannelId, bool), (override));

  MOCK_METHOD(VoipResult,
              GetInputVolumeInfo,
              (ChannelId, VolumeInfo&),
              (override));
  MOCK_METHOD(VoipResult,
              GetOutputVolumeInfo,
              (ChannelId, VolumeInfo&),
              (override));
};

class MockVoipEngine : public VoipEngine {
 public:
  VoipBase& Base() override { return base_; }
  VoipNetwork& Network() override { return network_; }
  VoipCodec& Codec() override { return codec_; }
  VoipDtmf& Dtmf() override { return dtmf_; }
  VoipStatistics& Statistics() override { return statistics_; }
  VoipVolumeControl& VolumeControl() override { return volume_; }

  // Direct access to underlying members are required for testing.
  MockVoipBase base_;
  MockVoipNetwork network_;
  MockVoipCodec codec_;
  MockVoipDtmf dtmf_;
  MockVoipStatistics statistics_;
  MockVoipVolumeControl volume_;
};

}  // namespace webrtc

#endif  // API_VOIP_TEST_MOCK_VOIP_ENGINE_H_