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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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72 | 72 |
73 const webrtc::AudioReceiveStream::Config& GetConfig() const; | 73 const webrtc::AudioReceiveStream::Config& GetConfig() const; |
74 void SetStats(const webrtc::AudioReceiveStream::Stats& stats); | 74 void SetStats(const webrtc::AudioReceiveStream::Stats& stats); |
75 int received_packets() const { return received_packets_; } | 75 int received_packets() const { return received_packets_; } |
76 bool VerifyLastPacket(const uint8_t* data, size_t length) const; | 76 bool VerifyLastPacket(const uint8_t* data, size_t length) const; |
77 const webrtc::AudioSinkInterface* sink() const { return sink_.get(); } | 77 const webrtc::AudioSinkInterface* sink() const { return sink_.get(); } |
78 float gain() const { return gain_; } | 78 float gain() const { return gain_; } |
79 bool DeliverRtp(const uint8_t* packet, | 79 bool DeliverRtp(const uint8_t* packet, |
80 size_t length, | 80 size_t length, |
81 const webrtc::PacketTime& packet_time); | 81 const webrtc::PacketTime& packet_time); |
| 82 bool started() const { return started_; } |
82 | 83 |
83 private: | 84 private: |
84 // webrtc::AudioReceiveStream implementation. | 85 // webrtc::AudioReceiveStream implementation. |
85 void Start() override {} | 86 void Start() override { started_ = true; } |
86 void Stop() override {} | 87 void Stop() override { started_ = false; } |
87 | 88 |
88 webrtc::AudioReceiveStream::Stats GetStats() const override; | 89 webrtc::AudioReceiveStream::Stats GetStats() const override; |
89 void SetSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) override; | 90 void SetSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) override; |
90 void SetGain(float gain) override; | 91 void SetGain(float gain) override; |
91 | 92 |
92 webrtc::AudioReceiveStream::Config config_; | 93 webrtc::AudioReceiveStream::Config config_; |
93 webrtc::AudioReceiveStream::Stats stats_; | 94 webrtc::AudioReceiveStream::Stats stats_; |
94 int received_packets_ = 0; | 95 int received_packets_ = 0; |
95 std::unique_ptr<webrtc::AudioSinkInterface> sink_; | 96 std::unique_ptr<webrtc::AudioSinkInterface> sink_; |
96 float gain_ = 1.0f; | 97 float gain_ = 1.0f; |
97 rtc::Buffer last_packet_; | 98 rtc::Buffer last_packet_; |
| 99 bool started_ = false; |
98 }; | 100 }; |
99 | 101 |
100 class FakeVideoSendStream final : public webrtc::VideoSendStream, | 102 class FakeVideoSendStream final : public webrtc::VideoSendStream, |
101 public webrtc::VideoCaptureInput { | 103 public webrtc::VideoCaptureInput { |
102 public: | 104 public: |
103 FakeVideoSendStream(const webrtc::VideoSendStream::Config& config, | 105 FakeVideoSendStream(const webrtc::VideoSendStream::Config& config, |
104 const webrtc::VideoEncoderConfig& encoder_config); | 106 const webrtc::VideoEncoderConfig& encoder_config); |
105 webrtc::VideoSendStream::Config GetConfig() const; | 107 webrtc::VideoSendStream::Config GetConfig() const; |
106 webrtc::VideoEncoderConfig GetEncoderConfig() const; | 108 webrtc::VideoEncoderConfig GetEncoderConfig() const; |
107 std::vector<webrtc::VideoStream> GetVideoStreams(); | 109 std::vector<webrtc::VideoStream> GetVideoStreams(); |
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245 std::vector<FakeAudioSendStream*> audio_send_streams_; | 247 std::vector<FakeAudioSendStream*> audio_send_streams_; |
246 std::vector<FakeVideoReceiveStream*> video_receive_streams_; | 248 std::vector<FakeVideoReceiveStream*> video_receive_streams_; |
247 std::vector<FakeAudioReceiveStream*> audio_receive_streams_; | 249 std::vector<FakeAudioReceiveStream*> audio_receive_streams_; |
248 | 250 |
249 int num_created_send_streams_; | 251 int num_created_send_streams_; |
250 int num_created_receive_streams_; | 252 int num_created_receive_streams_; |
251 }; | 253 }; |
252 | 254 |
253 } // namespace cricket | 255 } // namespace cricket |
254 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_ | 256 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_ |
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