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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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40 }; | 40 }; |
41 | 41 |
42 explicit FakeAudioSendStream(const webrtc::AudioSendStream::Config& config); | 42 explicit FakeAudioSendStream(const webrtc::AudioSendStream::Config& config); |
43 | 43 |
44 const webrtc::AudioSendStream::Config& GetConfig() const; | 44 const webrtc::AudioSendStream::Config& GetConfig() const; |
45 void SetStats(const webrtc::AudioSendStream::Stats& stats); | 45 void SetStats(const webrtc::AudioSendStream::Stats& stats); |
46 TelephoneEvent GetLatestTelephoneEvent() const; | 46 TelephoneEvent GetLatestTelephoneEvent() const; |
47 bool IsSending() const { return sending_; } | 47 bool IsSending() const { return sending_; } |
48 | 48 |
49 private: | 49 private: |
50 // webrtc::SendStream implementation. | 50 // webrtc::AudioSendStream implementation. |
51 void Start() override { sending_ = true; } | 51 void Start() override { sending_ = true; } |
52 void Stop() override { sending_ = false; } | 52 void Stop() override { sending_ = false; } |
53 void SignalNetworkState(webrtc::NetworkState state) override {} | |
54 bool DeliverRtcp(const uint8_t* packet, size_t length) override { | 53 bool DeliverRtcp(const uint8_t* packet, size_t length) override { |
The Sun (google.com)
2016/04/27 19:39:46
Unused, remove.
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55 return true; | 54 return true; |
56 } | 55 } |
57 | 56 |
58 // webrtc::AudioSendStream implementation. | |
59 bool SendTelephoneEvent(int payload_type, int event, | 57 bool SendTelephoneEvent(int payload_type, int event, |
60 int duration_ms) override; | 58 int duration_ms) override; |
61 webrtc::AudioSendStream::Stats GetStats() const override; | 59 webrtc::AudioSendStream::Stats GetStats() const override; |
62 | 60 |
63 TelephoneEvent latest_telephone_event_; | 61 TelephoneEvent latest_telephone_event_; |
64 webrtc::AudioSendStream::Config config_; | 62 webrtc::AudioSendStream::Config config_; |
65 webrtc::AudioSendStream::Stats stats_; | 63 webrtc::AudioSendStream::Stats stats_; |
66 bool sending_ = false; | 64 bool sending_ = false; |
67 }; | 65 }; |
68 | 66 |
69 class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream { | 67 class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream { |
70 public: | 68 public: |
71 explicit FakeAudioReceiveStream( | 69 explicit FakeAudioReceiveStream( |
72 const webrtc::AudioReceiveStream::Config& config); | 70 const webrtc::AudioReceiveStream::Config& config); |
73 | 71 |
74 const webrtc::AudioReceiveStream::Config& GetConfig() const; | 72 const webrtc::AudioReceiveStream::Config& GetConfig() const; |
75 void SetStats(const webrtc::AudioReceiveStream::Stats& stats); | 73 void SetStats(const webrtc::AudioReceiveStream::Stats& stats); |
76 int received_packets() const { return received_packets_; } | 74 int received_packets() const { return received_packets_; } |
77 void IncrementReceivedPackets(); | 75 void IncrementReceivedPackets(); |
78 const webrtc::AudioSinkInterface* sink() const { return sink_.get(); } | 76 const webrtc::AudioSinkInterface* sink() const { return sink_.get(); } |
79 | 77 |
80 private: | 78 private: |
81 // webrtc::ReceiveStream implementation. | 79 // webrtc::AudioReceiveStream implementation. |
82 void Start() override {} | 80 void Start() override {} |
83 void Stop() override {} | 81 void Stop() override {} |
84 void SignalNetworkState(webrtc::NetworkState state) override {} | |
85 bool DeliverRtcp(const uint8_t* packet, size_t length) override { | 82 bool DeliverRtcp(const uint8_t* packet, size_t length) override { |
The Sun (google.com)
2016/04/27 19:39:46
etc
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86 return true; | 83 return true; |
87 } | 84 } |
88 bool DeliverRtp(const uint8_t* packet, | 85 bool DeliverRtp(const uint8_t* packet, |
89 size_t length, | 86 size_t length, |
90 const webrtc::PacketTime& packet_time) override { | 87 const webrtc::PacketTime& packet_time) override { |
91 return true; | 88 return true; |
92 } | 89 } |
93 | 90 |
94 // webrtc::AudioReceiveStream implementation. | |
95 webrtc::AudioReceiveStream::Stats GetStats() const override; | 91 webrtc::AudioReceiveStream::Stats GetStats() const override; |
96 void SetSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) override; | 92 void SetSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) override; |
97 | 93 |
98 webrtc::AudioReceiveStream::Config config_; | 94 webrtc::AudioReceiveStream::Config config_; |
99 webrtc::AudioReceiveStream::Stats stats_; | 95 webrtc::AudioReceiveStream::Stats stats_; |
100 int received_packets_; | 96 int received_packets_; |
101 std::unique_ptr<webrtc::AudioSinkInterface> sink_; | 97 std::unique_ptr<webrtc::AudioSinkInterface> sink_; |
102 }; | 98 }; |
103 | 99 |
104 class FakeVideoSendStream final : public webrtc::VideoSendStream, | 100 class FakeVideoSendStream final : public webrtc::VideoSendStream, |
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119 int GetLastHeight() const; | 115 int GetLastHeight() const; |
120 int64_t GetLastTimestamp() const; | 116 int64_t GetLastTimestamp() const; |
121 void SetStats(const webrtc::VideoSendStream::Stats& stats); | 117 void SetStats(const webrtc::VideoSendStream::Stats& stats); |
122 int num_encoder_reconfigurations() const { | 118 int num_encoder_reconfigurations() const { |
123 return num_encoder_reconfigurations_; | 119 return num_encoder_reconfigurations_; |
124 } | 120 } |
125 | 121 |
126 private: | 122 private: |
127 void IncomingCapturedFrame(const webrtc::VideoFrame& frame) override; | 123 void IncomingCapturedFrame(const webrtc::VideoFrame& frame) override; |
128 | 124 |
129 // webrtc::SendStream implementation. | 125 // webrtc::VideoSendStream implementation. |
130 void Start() override; | 126 void Start() override; |
131 void Stop() override; | 127 void Stop() override; |
132 void SignalNetworkState(webrtc::NetworkState state) override {} | |
133 bool DeliverRtcp(const uint8_t* packet, size_t length) override { | 128 bool DeliverRtcp(const uint8_t* packet, size_t length) override { |
The Sun (google.com)
2016/04/27 19:39:46
etc
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134 return true; | 129 return true; |
135 } | 130 } |
136 | |
137 // webrtc::VideoSendStream implementation. | |
138 webrtc::VideoSendStream::Stats GetStats() override; | 131 webrtc::VideoSendStream::Stats GetStats() override; |
139 void ReconfigureVideoEncoder( | 132 void ReconfigureVideoEncoder( |
140 const webrtc::VideoEncoderConfig& config) override; | 133 const webrtc::VideoEncoderConfig& config) override; |
141 webrtc::VideoCaptureInput* Input() override; | 134 webrtc::VideoCaptureInput* Input() override; |
142 | 135 |
143 bool sending_; | 136 bool sending_; |
144 webrtc::VideoSendStream::Config config_; | 137 webrtc::VideoSendStream::Config config_; |
145 webrtc::VideoEncoderConfig encoder_config_; | 138 webrtc::VideoEncoderConfig encoder_config_; |
146 bool codec_settings_set_; | 139 bool codec_settings_set_; |
147 union VpxSettings { | 140 union VpxSettings { |
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161 | 154 |
162 webrtc::VideoReceiveStream::Config GetConfig(); | 155 webrtc::VideoReceiveStream::Config GetConfig(); |
163 | 156 |
164 bool IsReceiving() const; | 157 bool IsReceiving() const; |
165 | 158 |
166 void InjectFrame(const webrtc::VideoFrame& frame); | 159 void InjectFrame(const webrtc::VideoFrame& frame); |
167 | 160 |
168 void SetStats(const webrtc::VideoReceiveStream::Stats& stats); | 161 void SetStats(const webrtc::VideoReceiveStream::Stats& stats); |
169 | 162 |
170 private: | 163 private: |
171 // webrtc::ReceiveStream implementation. | 164 // webrtc::VideoReceiveStream implementation. |
172 void Start() override; | 165 void Start() override; |
173 void Stop() override; | 166 void Stop() override; |
174 void SignalNetworkState(webrtc::NetworkState state) override {} | |
175 bool DeliverRtcp(const uint8_t* packet, size_t length) override { | 167 bool DeliverRtcp(const uint8_t* packet, size_t length) override { |
The Sun (google.com)
2016/04/27 19:39:46
etc
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176 return true; | 168 return true; |
177 } | 169 } |
178 bool DeliverRtp(const uint8_t* packet, | 170 bool DeliverRtp(const uint8_t* packet, |
179 size_t length, | 171 size_t length, |
180 const webrtc::PacketTime& packet_time) override { | 172 const webrtc::PacketTime& packet_time) override { |
181 return true; | 173 return true; |
182 } | 174 } |
183 | 175 |
184 // webrtc::VideoReceiveStream implementation. | |
185 webrtc::VideoReceiveStream::Stats GetStats() const override; | 176 webrtc::VideoReceiveStream::Stats GetStats() const override; |
186 | 177 |
187 webrtc::VideoReceiveStream::Config config_; | 178 webrtc::VideoReceiveStream::Config config_; |
188 bool receiving_; | 179 bool receiving_; |
189 webrtc::VideoReceiveStream::Stats stats_; | 180 webrtc::VideoReceiveStream::Stats stats_; |
190 }; | 181 }; |
191 | 182 |
192 class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver { | 183 class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver { |
193 public: | 184 public: |
194 explicit FakeCall(const webrtc::Call::Config& config); | 185 explicit FakeCall(const webrtc::Call::Config& config); |
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254 std::vector<FakeAudioSendStream*> audio_send_streams_; | 245 std::vector<FakeAudioSendStream*> audio_send_streams_; |
255 std::vector<FakeVideoReceiveStream*> video_receive_streams_; | 246 std::vector<FakeVideoReceiveStream*> video_receive_streams_; |
256 std::vector<FakeAudioReceiveStream*> audio_receive_streams_; | 247 std::vector<FakeAudioReceiveStream*> audio_receive_streams_; |
257 | 248 |
258 int num_created_send_streams_; | 249 int num_created_send_streams_; |
259 int num_created_receive_streams_; | 250 int num_created_receive_streams_; |
260 }; | 251 }; |
261 | 252 |
262 } // namespace cricket | 253 } // namespace cricket |
263 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_ | 254 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_ |
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