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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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99 return true; | 99 return true; |
100 } | 100 } |
101 bool DeliverRtp(const uint8_t* packet, | 101 bool DeliverRtp(const uint8_t* packet, |
102 size_t length, | 102 size_t length, |
103 const webrtc::PacketTime& packet_time) override { | 103 const webrtc::PacketTime& packet_time) override { |
104 return true; | 104 return true; |
105 } | 105 } |
106 | 106 |
107 // webrtc::AudioReceiveStream implementation. | 107 // webrtc::AudioReceiveStream implementation. |
108 webrtc::AudioReceiveStream::Stats GetStats() const override; | 108 webrtc::AudioReceiveStream::Stats GetStats() const override; |
| 109 void SetSink(rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) override; |
109 | 110 |
110 webrtc::AudioReceiveStream::Config config_; | 111 webrtc::AudioReceiveStream::Config config_; |
111 webrtc::AudioReceiveStream::Stats stats_; | 112 webrtc::AudioReceiveStream::Stats stats_; |
112 int received_packets_; | 113 int received_packets_; |
| 114 rtc::scoped_ptr<webrtc::AudioSinkInterface> sink_; |
113 }; | 115 }; |
114 | 116 |
115 class FakeVideoSendStream final : public webrtc::VideoSendStream, | 117 class FakeVideoSendStream final : public webrtc::VideoSendStream, |
116 public webrtc::VideoCaptureInput { | 118 public webrtc::VideoCaptureInput { |
117 public: | 119 public: |
118 FakeVideoSendStream(const webrtc::VideoSendStream::Config& config, | 120 FakeVideoSendStream(const webrtc::VideoSendStream::Config& config, |
119 const webrtc::VideoEncoderConfig& encoder_config); | 121 const webrtc::VideoEncoderConfig& encoder_config); |
120 webrtc::VideoSendStream::Config GetConfig() const; | 122 webrtc::VideoSendStream::Config GetConfig() const; |
121 webrtc::VideoEncoderConfig GetEncoderConfig() const; | 123 webrtc::VideoEncoderConfig GetEncoderConfig() const; |
122 std::vector<webrtc::VideoStream> GetVideoStreams(); | 124 std::vector<webrtc::VideoStream> GetVideoStreams(); |
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257 std::vector<FakeAudioSendStream*> audio_send_streams_; | 259 std::vector<FakeAudioSendStream*> audio_send_streams_; |
258 std::vector<FakeVideoReceiveStream*> video_receive_streams_; | 260 std::vector<FakeVideoReceiveStream*> video_receive_streams_; |
259 std::vector<FakeAudioReceiveStream*> audio_receive_streams_; | 261 std::vector<FakeAudioReceiveStream*> audio_receive_streams_; |
260 | 262 |
261 int num_created_send_streams_; | 263 int num_created_send_streams_; |
262 int num_created_receive_streams_; | 264 int num_created_receive_streams_; |
263 }; | 265 }; |
264 | 266 |
265 } // namespace cricket | 267 } // namespace cricket |
266 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_ | 268 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_ |
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