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1 /* | 1 /* |
2 * libjingle | 2 * libjingle |
3 * Copyright 2015 Google Inc. | 3 * Copyright 2015 Google Inc. |
4 * | 4 * |
5 * Redistribution and use in source and binary forms, with or without | 5 * Redistribution and use in source and binary forms, with or without |
6 * modification, are permitted provided that the following conditions are met: | 6 * modification, are permitted provided that the following conditions are met: |
7 * | 7 * |
8 * 1. Redistributions of source code must retain the above copyright notice, | 8 * 1. Redistributions of source code must retain the above copyright notice, |
9 * this list of conditions and the following disclaimer. | 9 * this list of conditions and the following disclaimer. |
10 * 2. Redistributions in binary form must reproduce the above copyright notice, | 10 * 2. Redistributions in binary form must reproduce the above copyright notice, |
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82 | 82 |
83 class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream { | 83 class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream { |
84 public: | 84 public: |
85 explicit FakeAudioReceiveStream( | 85 explicit FakeAudioReceiveStream( |
86 const webrtc::AudioReceiveStream::Config& config); | 86 const webrtc::AudioReceiveStream::Config& config); |
87 | 87 |
88 const webrtc::AudioReceiveStream::Config& GetConfig() const; | 88 const webrtc::AudioReceiveStream::Config& GetConfig() const; |
89 void SetStats(const webrtc::AudioReceiveStream::Stats& stats); | 89 void SetStats(const webrtc::AudioReceiveStream::Stats& stats); |
90 int received_packets() const { return received_packets_; } | 90 int received_packets() const { return received_packets_; } |
91 void IncrementReceivedPackets(); | 91 void IncrementReceivedPackets(); |
92 rtc::scoped_refptr<webrtc::AudioSinkInterface> sink() const { return sink_; } | |
the sun
2016/01/11 10:38:27
super nit: could return a const &
| |
92 | 93 |
93 private: | 94 private: |
94 // webrtc::ReceiveStream implementation. | 95 // webrtc::ReceiveStream implementation. |
95 void Start() override {} | 96 void Start() override {} |
96 void Stop() override {} | 97 void Stop() override {} |
97 void SignalNetworkState(webrtc::NetworkState state) override {} | 98 void SignalNetworkState(webrtc::NetworkState state) override {} |
98 bool DeliverRtcp(const uint8_t* packet, size_t length) override { | 99 bool DeliverRtcp(const uint8_t* packet, size_t length) override { |
99 return true; | 100 return true; |
100 } | 101 } |
101 bool DeliverRtp(const uint8_t* packet, | 102 bool DeliverRtp(const uint8_t* packet, |
102 size_t length, | 103 size_t length, |
103 const webrtc::PacketTime& packet_time) override { | 104 const webrtc::PacketTime& packet_time) override { |
104 return true; | 105 return true; |
105 } | 106 } |
106 | 107 |
107 // webrtc::AudioReceiveStream implementation. | 108 // webrtc::AudioReceiveStream implementation. |
108 webrtc::AudioReceiveStream::Stats GetStats() const override; | 109 webrtc::AudioReceiveStream::Stats GetStats() const override; |
109 void SetSink(rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) override; | 110 void SetSink( |
111 const rtc::scoped_refptr<webrtc::AudioSinkInterface>& sink) override; | |
110 | 112 |
111 webrtc::AudioReceiveStream::Config config_; | 113 webrtc::AudioReceiveStream::Config config_; |
112 webrtc::AudioReceiveStream::Stats stats_; | 114 webrtc::AudioReceiveStream::Stats stats_; |
113 int received_packets_; | 115 int received_packets_; |
114 rtc::scoped_ptr<webrtc::AudioSinkInterface> sink_; | 116 rtc::scoped_refptr<webrtc::AudioSinkInterface> sink_; |
115 }; | 117 }; |
116 | 118 |
117 class FakeVideoSendStream final : public webrtc::VideoSendStream, | 119 class FakeVideoSendStream final : public webrtc::VideoSendStream, |
118 public webrtc::VideoCaptureInput { | 120 public webrtc::VideoCaptureInput { |
119 public: | 121 public: |
120 FakeVideoSendStream(const webrtc::VideoSendStream::Config& config, | 122 FakeVideoSendStream(const webrtc::VideoSendStream::Config& config, |
121 const webrtc::VideoEncoderConfig& encoder_config); | 123 const webrtc::VideoEncoderConfig& encoder_config); |
122 webrtc::VideoSendStream::Config GetConfig() const; | 124 webrtc::VideoSendStream::Config GetConfig() const; |
123 webrtc::VideoEncoderConfig GetEncoderConfig() const; | 125 webrtc::VideoEncoderConfig GetEncoderConfig() const; |
124 std::vector<webrtc::VideoStream> GetVideoStreams(); | 126 std::vector<webrtc::VideoStream> GetVideoStreams(); |
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259 std::vector<FakeAudioSendStream*> audio_send_streams_; | 261 std::vector<FakeAudioSendStream*> audio_send_streams_; |
260 std::vector<FakeVideoReceiveStream*> video_receive_streams_; | 262 std::vector<FakeVideoReceiveStream*> video_receive_streams_; |
261 std::vector<FakeAudioReceiveStream*> audio_receive_streams_; | 263 std::vector<FakeAudioReceiveStream*> audio_receive_streams_; |
262 | 264 |
263 int num_created_send_streams_; | 265 int num_created_send_streams_; |
264 int num_created_receive_streams_; | 266 int num_created_receive_streams_; |
265 }; | 267 }; |
266 | 268 |
267 } // namespace cricket | 269 } // namespace cricket |
268 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_ | 270 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_ |
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