OLD | NEW |
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 |
(...skipping 41 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
52 private: | 52 private: |
53 // webrtc::AudioSendStream implementation. | 53 // webrtc::AudioSendStream implementation. |
54 void Start() override { sending_ = true; } | 54 void Start() override { sending_ = true; } |
55 void Stop() override { sending_ = false; } | 55 void Stop() override { sending_ = false; } |
56 | 56 |
57 bool SendTelephoneEvent(int payload_type, int event, | 57 bool SendTelephoneEvent(int payload_type, int event, |
58 int duration_ms) override; | 58 int duration_ms) override; |
59 void SetMuted(bool muted) override; | 59 void SetMuted(bool muted) override; |
60 webrtc::AudioSendStream::Stats GetStats() const override; | 60 webrtc::AudioSendStream::Stats GetStats() const override; |
61 | 61 |
| 62 bool EnableAudioNetworkAdaptor(const std::string& config_string) override; |
| 63 |
| 64 void DisableAudioNetworkAdaptor() override; |
| 65 |
| 66 void SetReceiverFrameLengthRange(int min_frame_length_ms, |
| 67 int max_frame_length_ms) override; |
| 68 |
62 TelephoneEvent latest_telephone_event_; | 69 TelephoneEvent latest_telephone_event_; |
63 webrtc::AudioSendStream::Config config_; | 70 webrtc::AudioSendStream::Config config_; |
64 webrtc::AudioSendStream::Stats stats_; | 71 webrtc::AudioSendStream::Stats stats_; |
65 bool sending_ = false; | 72 bool sending_ = false; |
66 bool muted_ = false; | 73 bool muted_ = false; |
67 }; | 74 }; |
68 | 75 |
69 class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream { | 76 class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream { |
70 public: | 77 public: |
71 explicit FakeAudioReceiveStream( | 78 explicit FakeAudioReceiveStream( |
(...skipping 187 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
259 std::vector<FakeAudioSendStream*> audio_send_streams_; | 266 std::vector<FakeAudioSendStream*> audio_send_streams_; |
260 std::vector<FakeVideoReceiveStream*> video_receive_streams_; | 267 std::vector<FakeVideoReceiveStream*> video_receive_streams_; |
261 std::vector<FakeAudioReceiveStream*> audio_receive_streams_; | 268 std::vector<FakeAudioReceiveStream*> audio_receive_streams_; |
262 | 269 |
263 int num_created_send_streams_; | 270 int num_created_send_streams_; |
264 int num_created_receive_streams_; | 271 int num_created_receive_streams_; |
265 }; | 272 }; |
266 | 273 |
267 } // namespace cricket | 274 } // namespace cricket |
268 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_ | 275 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_ |
OLD | NEW |