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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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63 WEBRTC_STUB(Initialize, ( | 63 WEBRTC_STUB(Initialize, ( |
64 int input_sample_rate_hz, | 64 int input_sample_rate_hz, |
65 int output_sample_rate_hz, | 65 int output_sample_rate_hz, |
66 int reverse_sample_rate_hz, | 66 int reverse_sample_rate_hz, |
67 webrtc::AudioProcessing::ChannelLayout input_layout, | 67 webrtc::AudioProcessing::ChannelLayout input_layout, |
68 webrtc::AudioProcessing::ChannelLayout output_layout, | 68 webrtc::AudioProcessing::ChannelLayout output_layout, |
69 webrtc::AudioProcessing::ChannelLayout reverse_layout)); | 69 webrtc::AudioProcessing::ChannelLayout reverse_layout)); |
70 WEBRTC_STUB(Initialize, ( | 70 WEBRTC_STUB(Initialize, ( |
71 const webrtc::ProcessingConfig& processing_config)); | 71 const webrtc::ProcessingConfig& processing_config)); |
72 | 72 |
73 WEBRTC_VOID_STUB(ApplyConfig, (const AudioProcessing::Config& config)); | |
74 WEBRTC_VOID_FUNC(SetExtraOptions, (const webrtc::Config& config)) { | 73 WEBRTC_VOID_FUNC(SetExtraOptions, (const webrtc::Config& config)) { |
75 experimental_ns_enabled_ = config.Get<webrtc::ExperimentalNs>().enabled; | 74 experimental_ns_enabled_ = config.Get<webrtc::ExperimentalNs>().enabled; |
76 } | 75 } |
77 | 76 |
78 WEBRTC_STUB_CONST(proc_sample_rate_hz, ()); | 77 WEBRTC_STUB_CONST(proc_sample_rate_hz, ()); |
79 WEBRTC_STUB_CONST(proc_split_sample_rate_hz, ()); | 78 WEBRTC_STUB_CONST(proc_split_sample_rate_hz, ()); |
80 size_t num_input_channels() const override { return 0; } | 79 size_t num_input_channels() const override { return 0; } |
81 size_t num_proc_channels() const override { return 0; } | 80 size_t num_proc_channels() const override { return 0; } |
82 size_t num_output_channels() const override { return 0; } | 81 size_t num_output_channels() const override { return 0; } |
83 size_t num_reverse_channels() const override { return 0; } | 82 size_t num_reverse_channels() const override { return 0; } |
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571 webrtc::AecmModes aecm_mode_ = webrtc::kAecmSpeakerphone; | 570 webrtc::AecmModes aecm_mode_ = webrtc::kAecmSpeakerphone; |
572 webrtc::NsModes ns_mode_ = webrtc::kNsDefault; | 571 webrtc::NsModes ns_mode_ = webrtc::kNsDefault; |
573 webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault; | 572 webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault; |
574 webrtc::AgcConfig agc_config_; | 573 webrtc::AgcConfig agc_config_; |
575 FakeAudioProcessing audio_processing_; | 574 FakeAudioProcessing audio_processing_; |
576 }; | 575 }; |
577 | 576 |
578 } // namespace cricket | 577 } // namespace cricket |
579 | 578 |
580 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ | 579 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ |
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