35 lines
		
	
	
		
			1.3 KiB
		
	
	
	
		
			C++
		
	
	
	
			
		
		
	
	
			35 lines
		
	
	
		
			1.3 KiB
		
	
	
	
		
			C++
		
	
	
	
| /*
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|  *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
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|  *
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|  *  Use of this source code is governed by a BSD-style license
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|  *  that can be found in the LICENSE file in the root of the source
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|  *  tree. An additional intellectual property rights grant can be found
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|  *  in the file PATENTS.  All contributing project authors may
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|  *  be found in the AUTHORS file in the root of the source tree.
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|  */
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| 
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| #ifndef MODULES_AUDIO_CODING_ACM2_ACM_REMIXING_H_
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| #define MODULES_AUDIO_CODING_ACM2_ACM_REMIXING_H_
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| 
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| #include <vector>
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| 
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| #include "api/audio/audio_frame.h"
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| 
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| namespace webrtc {
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| 
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| // Stereo-to-mono downmixing. The length of the output must equal to the number
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| // of samples per channel in the input.
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| void DownMixFrame(const AudioFrame& input, rtc::ArrayView<int16_t> output);
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| 
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| // Remixes the interleaved input frame to an interleaved output data vector. The
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| // remixed data replaces the data in the output vector which is resized if
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| // needed. The remixing supports any combination of input and output channels,
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| // as well as any number of samples per channel.
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| void ReMixFrame(const AudioFrame& input,
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|                 size_t num_output_channels,
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|                 std::vector<int16_t>* output);
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| 
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| }  // namespace webrtc
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| 
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| #endif  // MODULES_AUDIO_CODING_ACM2_ACM_REMIXING_H_
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