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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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90 | 90 |
91 private: | 91 private: |
92 // webrtc::AudioReceiveStream implementation. | 92 // webrtc::AudioReceiveStream implementation. |
93 void Start() override { started_ = true; } | 93 void Start() override { started_ = true; } |
94 void Stop() override { started_ = false; } | 94 void Stop() override { started_ = false; } |
95 | 95 |
96 webrtc::AudioReceiveStream::Stats GetStats() const override; | 96 webrtc::AudioReceiveStream::Stats GetStats() const override; |
97 int GetOutputLevel() const override { return 0; } | 97 int GetOutputLevel() const override { return 0; } |
98 void SetSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) override; | 98 void SetSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) override; |
99 void SetGain(float gain) override; | 99 void SetGain(float gain) override; |
100 std::vector<webrtc::RtpSource> GetSources() const override { | |
101 return std::vector<webrtc::RtpSource>(); | |
102 } | |
103 | 100 |
104 int id_ = -1; | 101 int id_ = -1; |
105 webrtc::AudioReceiveStream::Config config_; | 102 webrtc::AudioReceiveStream::Config config_; |
106 webrtc::AudioReceiveStream::Stats stats_; | 103 webrtc::AudioReceiveStream::Stats stats_; |
107 int received_packets_ = 0; | 104 int received_packets_ = 0; |
108 std::unique_ptr<webrtc::AudioSinkInterface> sink_; | 105 std::unique_ptr<webrtc::AudioSinkInterface> sink_; |
109 float gain_ = 1.0f; | 106 float gain_ = 1.0f; |
110 rtc::Buffer last_packet_; | 107 rtc::Buffer last_packet_; |
111 bool started_ = false; | 108 bool started_ = false; |
112 }; | 109 }; |
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314 | 311 |
315 int num_created_send_streams_; | 312 int num_created_send_streams_; |
316 int num_created_receive_streams_; | 313 int num_created_receive_streams_; |
317 | 314 |
318 int audio_transport_overhead_; | 315 int audio_transport_overhead_; |
319 int video_transport_overhead_; | 316 int video_transport_overhead_; |
320 }; | 317 }; |
321 | 318 |
322 } // namespace cricket | 319 } // namespace cricket |
323 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_ | 320 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_ |
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