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1 /* | |
2 * libjingle | |
3 * Copyright 2015 Google Inc. | |
4 * | |
5 * Redistribution and use in source and binary forms, with or without | |
6 * modification, are permitted provided that the following conditions are met: | |
7 * | |
8 * 1. Redistributions of source code must retain the above copyright notice, | |
9 * this list of conditions and the following disclaimer. | |
10 * 2. Redistributions in binary form must reproduce the above copyright notice, | |
11 * this list of conditions and the following disclaimer in the documentation | |
12 * and/or other materials provided with the distribution. | |
13 * 3. The name of the author may not be used to endorse or promote products | |
14 * derived from this software without specific prior written permission. | |
15 * | |
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED | |
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF | |
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO | |
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, | |
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, | |
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; | |
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, | |
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR | |
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF | |
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. | |
26 */ | |
27 | |
28 // This file contains fake implementations, for use in unit tests, of the | |
29 // following classes: | |
30 // | |
31 // webrtc::Call | |
32 // webrtc::AudioSendStream | |
33 // webrtc::AudioReceiveStream | |
34 // webrtc::VideoSendStream | |
35 // webrtc::VideoReceiveStream | |
36 | |
37 #ifndef TALK_MEDIA_WEBRTC_FAKEWEBRTCCALL_H_ | |
38 #define TALK_MEDIA_WEBRTC_FAKEWEBRTCCALL_H_ | |
39 | |
40 #include <vector> | |
41 | |
42 #include "webrtc/call.h" | |
43 #include "webrtc/audio_receive_stream.h" | |
44 #include "webrtc/audio_send_stream.h" | |
45 #include "webrtc/video_frame.h" | |
46 #include "webrtc/video_receive_stream.h" | |
47 #include "webrtc/video_send_stream.h" | |
48 | |
49 namespace cricket { | |
50 class FakeAudioSendStream final : public webrtc::AudioSendStream { | |
51 public: | |
52 struct TelephoneEvent { | |
53 int payload_type = -1; | |
54 uint8_t event_code = 0; | |
55 uint32_t duration_ms = 0; | |
56 }; | |
57 | |
58 explicit FakeAudioSendStream(const webrtc::AudioSendStream::Config& config); | |
59 | |
60 const webrtc::AudioSendStream::Config& GetConfig() const; | |
61 void SetStats(const webrtc::AudioSendStream::Stats& stats); | |
62 TelephoneEvent GetLatestTelephoneEvent() const; | |
63 | |
64 private: | |
65 // webrtc::SendStream implementation. | |
66 void Start() override {} | |
67 void Stop() override {} | |
68 void SignalNetworkState(webrtc::NetworkState state) override {} | |
69 bool DeliverRtcp(const uint8_t* packet, size_t length) override { | |
70 return true; | |
71 } | |
72 | |
73 // webrtc::AudioSendStream implementation. | |
74 bool SendTelephoneEvent(int payload_type, uint8_t event, | |
75 uint32_t duration_ms) override; | |
76 webrtc::AudioSendStream::Stats GetStats() const override; | |
77 | |
78 TelephoneEvent latest_telephone_event_; | |
79 webrtc::AudioSendStream::Config config_; | |
80 webrtc::AudioSendStream::Stats stats_; | |
81 }; | |
82 | |
83 class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream { | |
84 public: | |
85 explicit FakeAudioReceiveStream( | |
86 const webrtc::AudioReceiveStream::Config& config); | |
87 | |
88 const webrtc::AudioReceiveStream::Config& GetConfig() const; | |
89 void SetStats(const webrtc::AudioReceiveStream::Stats& stats); | |
90 int received_packets() const { return received_packets_; } | |
91 void IncrementReceivedPackets(); | |
92 const webrtc::AudioSinkInterface* sink() const { return sink_.get(); } | |
93 | |
94 private: | |
95 // webrtc::ReceiveStream implementation. | |
96 void Start() override {} | |
97 void Stop() override {} | |
98 void SignalNetworkState(webrtc::NetworkState state) override {} | |
99 bool DeliverRtcp(const uint8_t* packet, size_t length) override { | |
100 return true; | |
101 } | |
102 bool DeliverRtp(const uint8_t* packet, | |
103 size_t length, | |
104 const webrtc::PacketTime& packet_time) override { | |
105 return true; | |
106 } | |
107 | |
108 // webrtc::AudioReceiveStream implementation. | |
109 webrtc::AudioReceiveStream::Stats GetStats() const override; | |
110 void SetSink(rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) override; | |
111 | |
112 webrtc::AudioReceiveStream::Config config_; | |
113 webrtc::AudioReceiveStream::Stats stats_; | |
114 int received_packets_; | |
115 rtc::scoped_ptr<webrtc::AudioSinkInterface> sink_; | |
116 }; | |
117 | |
118 class FakeVideoSendStream final : public webrtc::VideoSendStream, | |
119 public webrtc::VideoCaptureInput { | |
120 public: | |
121 FakeVideoSendStream(const webrtc::VideoSendStream::Config& config, | |
122 const webrtc::VideoEncoderConfig& encoder_config); | |
123 webrtc::VideoSendStream::Config GetConfig() const; | |
124 webrtc::VideoEncoderConfig GetEncoderConfig() const; | |
125 std::vector<webrtc::VideoStream> GetVideoStreams(); | |
126 | |
127 bool IsSending() const; | |
128 bool GetVp8Settings(webrtc::VideoCodecVP8* settings) const; | |
129 bool GetVp9Settings(webrtc::VideoCodecVP9* settings) const; | |
130 | |
131 int GetNumberOfSwappedFrames() const; | |
132 int GetLastWidth() const; | |
133 int GetLastHeight() const; | |
134 int64_t GetLastTimestamp() const; | |
135 void SetStats(const webrtc::VideoSendStream::Stats& stats); | |
136 | |
137 private: | |
138 void IncomingCapturedFrame(const webrtc::VideoFrame& frame) override; | |
139 | |
140 // webrtc::SendStream implementation. | |
141 void Start() override; | |
142 void Stop() override; | |
143 void SignalNetworkState(webrtc::NetworkState state) override {} | |
144 bool DeliverRtcp(const uint8_t* packet, size_t length) override { | |
145 return true; | |
146 } | |
147 | |
148 // webrtc::VideoSendStream implementation. | |
149 webrtc::VideoSendStream::Stats GetStats() override; | |
150 bool ReconfigureVideoEncoder( | |
151 const webrtc::VideoEncoderConfig& config) override; | |
152 webrtc::VideoCaptureInput* Input() override; | |
153 | |
154 bool sending_; | |
155 webrtc::VideoSendStream::Config config_; | |
156 webrtc::VideoEncoderConfig encoder_config_; | |
157 bool codec_settings_set_; | |
158 union VpxSettings { | |
159 webrtc::VideoCodecVP8 vp8; | |
160 webrtc::VideoCodecVP9 vp9; | |
161 } vpx_settings_; | |
162 int num_swapped_frames_; | |
163 webrtc::VideoFrame last_frame_; | |
164 webrtc::VideoSendStream::Stats stats_; | |
165 }; | |
166 | |
167 class FakeVideoReceiveStream final : public webrtc::VideoReceiveStream { | |
168 public: | |
169 explicit FakeVideoReceiveStream( | |
170 const webrtc::VideoReceiveStream::Config& config); | |
171 | |
172 webrtc::VideoReceiveStream::Config GetConfig(); | |
173 | |
174 bool IsReceiving() const; | |
175 | |
176 void InjectFrame(const webrtc::VideoFrame& frame, int time_to_render_ms); | |
177 | |
178 void SetStats(const webrtc::VideoReceiveStream::Stats& stats); | |
179 | |
180 private: | |
181 // webrtc::ReceiveStream implementation. | |
182 void Start() override; | |
183 void Stop() override; | |
184 void SignalNetworkState(webrtc::NetworkState state) override {} | |
185 bool DeliverRtcp(const uint8_t* packet, size_t length) override { | |
186 return true; | |
187 } | |
188 bool DeliverRtp(const uint8_t* packet, | |
189 size_t length, | |
190 const webrtc::PacketTime& packet_time) override { | |
191 return true; | |
192 } | |
193 | |
194 // webrtc::VideoReceiveStream implementation. | |
195 webrtc::VideoReceiveStream::Stats GetStats() const override; | |
196 | |
197 webrtc::VideoReceiveStream::Config config_; | |
198 bool receiving_; | |
199 webrtc::VideoReceiveStream::Stats stats_; | |
200 }; | |
201 | |
202 class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver { | |
203 public: | |
204 explicit FakeCall(const webrtc::Call::Config& config); | |
205 ~FakeCall() override; | |
206 | |
207 webrtc::Call::Config GetConfig() const; | |
208 const std::vector<FakeVideoSendStream*>& GetVideoSendStreams(); | |
209 const std::vector<FakeVideoReceiveStream*>& GetVideoReceiveStreams(); | |
210 | |
211 const std::vector<FakeAudioSendStream*>& GetAudioSendStreams(); | |
212 const FakeAudioSendStream* GetAudioSendStream(uint32_t ssrc); | |
213 const std::vector<FakeAudioReceiveStream*>& GetAudioReceiveStreams(); | |
214 const FakeAudioReceiveStream* GetAudioReceiveStream(uint32_t ssrc); | |
215 | |
216 rtc::SentPacket last_sent_packet() const { return last_sent_packet_; } | |
217 webrtc::NetworkState GetNetworkState() const; | |
218 int GetNumCreatedSendStreams() const; | |
219 int GetNumCreatedReceiveStreams() const; | |
220 void SetStats(const webrtc::Call::Stats& stats); | |
221 | |
222 private: | |
223 webrtc::AudioSendStream* CreateAudioSendStream( | |
224 const webrtc::AudioSendStream::Config& config) override; | |
225 void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override; | |
226 | |
227 webrtc::AudioReceiveStream* CreateAudioReceiveStream( | |
228 const webrtc::AudioReceiveStream::Config& config) override; | |
229 void DestroyAudioReceiveStream( | |
230 webrtc::AudioReceiveStream* receive_stream) override; | |
231 | |
232 webrtc::VideoSendStream* CreateVideoSendStream( | |
233 const webrtc::VideoSendStream::Config& config, | |
234 const webrtc::VideoEncoderConfig& encoder_config) override; | |
235 void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override; | |
236 | |
237 webrtc::VideoReceiveStream* CreateVideoReceiveStream( | |
238 const webrtc::VideoReceiveStream::Config& config) override; | |
239 void DestroyVideoReceiveStream( | |
240 webrtc::VideoReceiveStream* receive_stream) override; | |
241 webrtc::PacketReceiver* Receiver() override; | |
242 | |
243 DeliveryStatus DeliverPacket(webrtc::MediaType media_type, | |
244 const uint8_t* packet, | |
245 size_t length, | |
246 const webrtc::PacketTime& packet_time) override; | |
247 | |
248 webrtc::Call::Stats GetStats() const override; | |
249 | |
250 void SetBitrateConfig( | |
251 const webrtc::Call::Config::BitrateConfig& bitrate_config) override; | |
252 void SignalNetworkState(webrtc::NetworkState state) override; | |
253 void OnSentPacket(const rtc::SentPacket& sent_packet) override; | |
254 | |
255 webrtc::Call::Config config_; | |
256 webrtc::NetworkState network_state_; | |
257 rtc::SentPacket last_sent_packet_; | |
258 webrtc::Call::Stats stats_; | |
259 std::vector<FakeVideoSendStream*> video_send_streams_; | |
260 std::vector<FakeAudioSendStream*> audio_send_streams_; | |
261 std::vector<FakeVideoReceiveStream*> video_receive_streams_; | |
262 std::vector<FakeAudioReceiveStream*> audio_receive_streams_; | |
263 | |
264 int num_created_send_streams_; | |
265 int num_created_receive_streams_; | |
266 }; | |
267 | |
268 } // namespace cricket | |
269 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_ | |
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