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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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71 int input_sample_rate_hz, | 71 int input_sample_rate_hz, |
72 webrtc::AudioProcessing::ChannelLayout input_layout, | 72 webrtc::AudioProcessing::ChannelLayout input_layout, |
73 int output_sample_rate_hz, | 73 int output_sample_rate_hz, |
74 webrtc::AudioProcessing::ChannelLayout output_layout, | 74 webrtc::AudioProcessing::ChannelLayout output_layout, |
75 float* const* dest)); | 75 float* const* dest)); |
76 WEBRTC_STUB(ProcessStream, | 76 WEBRTC_STUB(ProcessStream, |
77 (const float* const* src, | 77 (const float* const* src, |
78 const webrtc::StreamConfig& input_config, | 78 const webrtc::StreamConfig& input_config, |
79 const webrtc::StreamConfig& output_config, | 79 const webrtc::StreamConfig& output_config, |
80 float* const* dest)); | 80 float* const* dest)); |
81 WEBRTC_STUB(AnalyzeReverseStream, (webrtc::AudioFrame* frame)); | |
82 WEBRTC_STUB(ProcessReverseStream, (webrtc::AudioFrame * frame)); | 81 WEBRTC_STUB(ProcessReverseStream, (webrtc::AudioFrame * frame)); |
83 WEBRTC_STUB(AnalyzeReverseStream, ( | 82 WEBRTC_STUB(AnalyzeReverseStream, ( |
84 const float* const* data, | 83 const float* const* data, |
85 size_t samples_per_channel, | 84 size_t samples_per_channel, |
86 int sample_rate_hz, | 85 int sample_rate_hz, |
87 webrtc::AudioProcessing::ChannelLayout layout)); | 86 webrtc::AudioProcessing::ChannelLayout layout)); |
88 WEBRTC_STUB(ProcessReverseStream, | 87 WEBRTC_STUB(ProcessReverseStream, |
89 (const float* const* src, | 88 (const float* const* src, |
90 const webrtc::StreamConfig& reverse_input_config, | 89 const webrtc::StreamConfig& reverse_input_config, |
91 const webrtc::StreamConfig& reverse_output_config, | 90 const webrtc::StreamConfig& reverse_output_config, |
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774 webrtc::VoiceEngineObserver* observer_; | 773 webrtc::VoiceEngineObserver* observer_; |
775 int playout_fail_channel_; | 774 int playout_fail_channel_; |
776 int recording_sample_rate_; | 775 int recording_sample_rate_; |
777 int playout_sample_rate_; | 776 int playout_sample_rate_; |
778 FakeAudioProcessing audio_processing_; | 777 FakeAudioProcessing audio_processing_; |
779 }; | 778 }; |
780 | 779 |
781 } // namespace cricket | 780 } // namespace cricket |
782 | 781 |
783 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ | 782 #endif // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_ |
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