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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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| 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( |
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| 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_ |
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