OLD | NEW |
1 /* | 1 /* |
2 * Copyright (c) 2010 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2010 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 102 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
113 WEBRTC_VOID_STUB(set_stream_key_pressed, (bool key_pressed)); | 113 WEBRTC_VOID_STUB(set_stream_key_pressed, (bool key_pressed)); |
114 WEBRTC_VOID_STUB(set_delay_offset_ms, (int offset)); | 114 WEBRTC_VOID_STUB(set_delay_offset_ms, (int offset)); |
115 WEBRTC_STUB_CONST(delay_offset_ms, ()); | 115 WEBRTC_STUB_CONST(delay_offset_ms, ()); |
116 WEBRTC_STUB(StartDebugRecording, | 116 WEBRTC_STUB(StartDebugRecording, |
117 (const char filename[kMaxFilenameSize], int64_t max_size_bytes)); | 117 (const char filename[kMaxFilenameSize], int64_t max_size_bytes)); |
118 WEBRTC_STUB(StartDebugRecording, (FILE * handle, int64_t max_size_bytes)); | 118 WEBRTC_STUB(StartDebugRecording, (FILE * handle, int64_t max_size_bytes)); |
119 WEBRTC_STUB(StartDebugRecording, (FILE * handle)); | 119 WEBRTC_STUB(StartDebugRecording, (FILE * handle)); |
120 WEBRTC_STUB(StartDebugRecordingForPlatformFile, (rtc::PlatformFile handle)); | 120 WEBRTC_STUB(StartDebugRecordingForPlatformFile, (rtc::PlatformFile handle)); |
121 WEBRTC_STUB(StopDebugRecording, ()); | 121 WEBRTC_STUB(StopDebugRecording, ()); |
122 WEBRTC_VOID_STUB(UpdateHistogramsOnCallEnd, ()); | 122 WEBRTC_VOID_STUB(UpdateHistogramsOnCallEnd, ()); |
| 123 AudioProcessing::AudioProcessingStatistics GetStatistics() const override { |
| 124 AudioProcessing::AudioProcessingStatistics stats; |
| 125 return stats; |
| 126 } |
123 webrtc::EchoCancellation* echo_cancellation() const override { return NULL; } | 127 webrtc::EchoCancellation* echo_cancellation() const override { return NULL; } |
124 webrtc::EchoControlMobile* echo_control_mobile() const override { | 128 webrtc::EchoControlMobile* echo_control_mobile() const override { |
125 return NULL; | 129 return NULL; |
126 } | 130 } |
127 webrtc::GainControl* gain_control() const override { return NULL; } | 131 webrtc::GainControl* gain_control() const override { return NULL; } |
128 webrtc::HighPassFilter* high_pass_filter() const override { return NULL; } | 132 webrtc::HighPassFilter* high_pass_filter() const override { return NULL; } |
129 webrtc::LevelEstimator* level_estimator() const override { return NULL; } | 133 webrtc::LevelEstimator* level_estimator() const override { return NULL; } |
130 webrtc::NoiseSuppression* noise_suppression() const override { return NULL; } | 134 webrtc::NoiseSuppression* noise_suppression() const override { return NULL; } |
131 webrtc::VoiceDetection* voice_detection() const override { return NULL; } | 135 webrtc::VoiceDetection* voice_detection() const override { return NULL; } |
132 | 136 |
(...skipping 428 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
561 webrtc::AecmModes aecm_mode_ = webrtc::kAecmSpeakerphone; | 565 webrtc::AecmModes aecm_mode_ = webrtc::kAecmSpeakerphone; |
562 webrtc::NsModes ns_mode_ = webrtc::kNsDefault; | 566 webrtc::NsModes ns_mode_ = webrtc::kNsDefault; |
563 webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault; | 567 webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault; |
564 webrtc::AgcConfig agc_config_; | 568 webrtc::AgcConfig agc_config_; |
565 FakeAudioProcessing audio_processing_; | 569 FakeAudioProcessing audio_processing_; |
566 }; | 570 }; |
567 | 571 |
568 } // namespace cricket | 572 } // namespace cricket |
569 | 573 |
570 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ | 574 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ |
OLD | NEW |