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Side by Side Diff: webrtc/media/engine/fakewebrtcvoiceengine.h

Issue 2433153003: New statistics interface for APM (Closed)
Patch Set: Review comments. Created 4 years, 1 month ago
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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
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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
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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_
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