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Side by Side Diff: webrtc/modules/audio_processing/audio_buffer.h

Issue 1253573005: Revert of Allow more than 2 input channels in AudioProcessing. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 5 months ago
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1 /* 1 /*
2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2011 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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105 void set_activity(AudioFrame::VADActivity activity); 105 void set_activity(AudioFrame::VADActivity activity);
106 AudioFrame::VADActivity activity() const; 106 AudioFrame::VADActivity activity() const;
107 107
108 // Use for int16 interleaved data. 108 // Use for int16 interleaved data.
109 void DeinterleaveFrom(AudioFrame* audioFrame); 109 void DeinterleaveFrom(AudioFrame* audioFrame);
110 // If |data_changed| is false, only the non-audio data members will be copied 110 // If |data_changed| is false, only the non-audio data members will be copied
111 // to |frame|. 111 // to |frame|.
112 void InterleaveTo(AudioFrame* frame, bool data_changed) const; 112 void InterleaveTo(AudioFrame* frame, bool data_changed) const;
113 113
114 // Use for float deinterleaved data. 114 // Use for float deinterleaved data.
115 void CopyFrom(const float* const* data, const StreamConfig& stream_config); 115 void CopyFrom(const float* const* data,
116 void CopyTo(const StreamConfig& stream_config, float* const* data); 116 int num_frames,
117 AudioProcessing::ChannelLayout layout);
118 void CopyTo(int num_frames,
119 AudioProcessing::ChannelLayout layout,
120 float* const* data);
117 void CopyLowPassToReference(); 121 void CopyLowPassToReference();
118 122
119 // Splits the signal into different bands. 123 // Splits the signal into different bands.
120 void SplitIntoFrequencyBands(); 124 void SplitIntoFrequencyBands();
121 // Recombine the different bands into one signal. 125 // Recombine the different bands into one signal.
122 void MergeFrequencyBands(); 126 void MergeFrequencyBands();
123 127
124 private: 128 private:
125 // Called from DeinterleaveFrom() and CopyFrom(). 129 // Called from DeinterleaveFrom() and CopyFrom().
126 void InitForNewData(); 130 void InitForNewData();
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153 rtc::scoped_ptr<ChannelBuffer<int16_t> > low_pass_reference_channels_; 157 rtc::scoped_ptr<ChannelBuffer<int16_t> > low_pass_reference_channels_;
154 rtc::scoped_ptr<IFChannelBuffer> input_buffer_; 158 rtc::scoped_ptr<IFChannelBuffer> input_buffer_;
155 rtc::scoped_ptr<ChannelBuffer<float> > process_buffer_; 159 rtc::scoped_ptr<ChannelBuffer<float> > process_buffer_;
156 ScopedVector<PushSincResampler> input_resamplers_; 160 ScopedVector<PushSincResampler> input_resamplers_;
157 ScopedVector<PushSincResampler> output_resamplers_; 161 ScopedVector<PushSincResampler> output_resamplers_;
158 }; 162 };
159 163
160 } // namespace webrtc 164 } // namespace webrtc
161 165
162 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_ 166 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_
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