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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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48 | 48 |
49 int id() const { return id_; } | 49 int id() const { return id_; } |
50 const webrtc::AudioSendStream::Config& GetConfig() const; | 50 const webrtc::AudioSendStream::Config& GetConfig() const; |
51 void SetStats(const webrtc::AudioSendStream::Stats& stats); | 51 void SetStats(const webrtc::AudioSendStream::Stats& stats); |
52 TelephoneEvent GetLatestTelephoneEvent() const; | 52 TelephoneEvent GetLatestTelephoneEvent() const; |
53 bool IsSending() const { return sending_; } | 53 bool IsSending() const { return sending_; } |
54 bool muted() const { return muted_; } | 54 bool muted() const { return muted_; } |
55 | 55 |
56 private: | 56 private: |
57 // webrtc::AudioSendStream implementation. | 57 // webrtc::AudioSendStream implementation. |
| 58 void Reconfigure(const webrtc::AudioSendStream::Config& config) override; |
| 59 |
58 void Start() override { sending_ = true; } | 60 void Start() override { sending_ = true; } |
59 void Stop() override { sending_ = false; } | 61 void Stop() override { sending_ = false; } |
60 | 62 |
61 bool SendTelephoneEvent(int payload_type, int payload_frequency, int event, | 63 bool SendTelephoneEvent(int payload_type, int payload_frequency, int event, |
62 int duration_ms) override; | 64 int duration_ms) override; |
63 void SetMuted(bool muted) override; | 65 void SetMuted(bool muted) override; |
64 webrtc::AudioSendStream::Stats GetStats() const override; | 66 webrtc::AudioSendStream::Stats GetStats() const override; |
65 | 67 |
66 int id_ = -1; | 68 int id_ = -1; |
67 TelephoneEvent latest_telephone_event_; | 69 TelephoneEvent latest_telephone_event_; |
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314 | 316 |
315 int num_created_send_streams_; | 317 int num_created_send_streams_; |
316 int num_created_receive_streams_; | 318 int num_created_receive_streams_; |
317 | 319 |
318 int audio_transport_overhead_; | 320 int audio_transport_overhead_; |
319 int video_transport_overhead_; | 321 int video_transport_overhead_; |
320 }; | 322 }; |
321 | 323 |
322 } // namespace cricket | 324 } // namespace cricket |
323 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_ | 325 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_ |
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