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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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| 39 int event_code = 0; | 39 int event_code = 0; |
| 40 int duration_ms = 0; | 40 int duration_ms = 0; |
| 41 }; | 41 }; |
| 42 | 42 |
| 43 explicit FakeAudioSendStream(const webrtc::AudioSendStream::Config& config); | 43 explicit FakeAudioSendStream(const webrtc::AudioSendStream::Config& config); |
| 44 | 44 |
| 45 const webrtc::AudioSendStream::Config& GetConfig() const; | 45 const webrtc::AudioSendStream::Config& GetConfig() const; |
| 46 void SetStats(const webrtc::AudioSendStream::Stats& stats); | 46 void SetStats(const webrtc::AudioSendStream::Stats& stats); |
| 47 TelephoneEvent GetLatestTelephoneEvent() const; | 47 TelephoneEvent GetLatestTelephoneEvent() const; |
| 48 bool IsSending() const { return sending_; } | 48 bool IsSending() const { return sending_; } |
| 49 bool muted() const { return muted_; } |
| 49 | 50 |
| 50 private: | 51 private: |
| 51 // webrtc::AudioSendStream implementation. | 52 // webrtc::AudioSendStream implementation. |
| 52 void Start() override { sending_ = true; } | 53 void Start() override { sending_ = true; } |
| 53 void Stop() override { sending_ = false; } | 54 void Stop() override { sending_ = false; } |
| 54 | 55 |
| 55 bool SendTelephoneEvent(int payload_type, int event, | 56 bool SendTelephoneEvent(int payload_type, int event, |
| 56 int duration_ms) override; | 57 int duration_ms) override; |
| 58 void SetMuted(bool muted) override; |
| 57 webrtc::AudioSendStream::Stats GetStats() const override; | 59 webrtc::AudioSendStream::Stats GetStats() const override; |
| 58 | 60 |
| 59 TelephoneEvent latest_telephone_event_; | 61 TelephoneEvent latest_telephone_event_; |
| 60 webrtc::AudioSendStream::Config config_; | 62 webrtc::AudioSendStream::Config config_; |
| 61 webrtc::AudioSendStream::Stats stats_; | 63 webrtc::AudioSendStream::Stats stats_; |
| 62 bool sending_ = false; | 64 bool sending_ = false; |
| 65 bool muted_ = false; |
| 63 }; | 66 }; |
| 64 | 67 |
| 65 class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream { | 68 class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream { |
| 66 public: | 69 public: |
| 67 explicit FakeAudioReceiveStream( | 70 explicit FakeAudioReceiveStream( |
| 68 const webrtc::AudioReceiveStream::Config& config); | 71 const webrtc::AudioReceiveStream::Config& config); |
| 69 | 72 |
| 70 const webrtc::AudioReceiveStream::Config& GetConfig() const; | 73 const webrtc::AudioReceiveStream::Config& GetConfig() const; |
| 71 void SetStats(const webrtc::AudioReceiveStream::Stats& stats); | 74 void SetStats(const webrtc::AudioReceiveStream::Stats& stats); |
| 72 int received_packets() const { return received_packets_; } | 75 int received_packets() const { return received_packets_; } |
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| 227 std::vector<FakeAudioSendStream*> audio_send_streams_; | 230 std::vector<FakeAudioSendStream*> audio_send_streams_; |
| 228 std::vector<FakeVideoReceiveStream*> video_receive_streams_; | 231 std::vector<FakeVideoReceiveStream*> video_receive_streams_; |
| 229 std::vector<FakeAudioReceiveStream*> audio_receive_streams_; | 232 std::vector<FakeAudioReceiveStream*> audio_receive_streams_; |
| 230 | 233 |
| 231 int num_created_send_streams_; | 234 int num_created_send_streams_; |
| 232 int num_created_receive_streams_; | 235 int num_created_receive_streams_; |
| 233 }; | 236 }; |
| 234 | 237 |
| 235 } // namespace cricket | 238 } // namespace cricket |
| 236 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_ | 239 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_ |
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