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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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87 size_t length, | 87 size_t length, |
88 const webrtc::PacketTime& packet_time); | 88 const webrtc::PacketTime& packet_time); |
89 bool started() const { return started_; } | 89 bool started() const { return started_; } |
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 void SetSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) override; | 98 void SetSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) override; |
98 void SetGain(float gain) override; | 99 void SetGain(float gain) override; |
99 | 100 |
100 int id_ = -1; | 101 int id_ = -1; |
101 webrtc::AudioReceiveStream::Config config_; | 102 webrtc::AudioReceiveStream::Config config_; |
102 webrtc::AudioReceiveStream::Stats stats_; | 103 webrtc::AudioReceiveStream::Stats stats_; |
103 int received_packets_ = 0; | 104 int received_packets_ = 0; |
104 std::unique_ptr<webrtc::AudioSinkInterface> sink_; | 105 std::unique_ptr<webrtc::AudioSinkInterface> sink_; |
105 float gain_ = 1.0f; | 106 float gain_ = 1.0f; |
106 rtc::Buffer last_packet_; | 107 rtc::Buffer last_packet_; |
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308 | 309 |
309 int num_created_send_streams_; | 310 int num_created_send_streams_; |
310 int num_created_receive_streams_; | 311 int num_created_receive_streams_; |
311 | 312 |
312 int audio_transport_overhead_; | 313 int audio_transport_overhead_; |
313 int video_transport_overhead_; | 314 int video_transport_overhead_; |
314 }; | 315 }; |
315 | 316 |
316 } // namespace cricket | 317 } // namespace cricket |
317 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_ | 318 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_ |
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