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			54 lines
		
	
	
		
			2.8 KiB
		
	
	
	
		
			C
		
	
		
		
			
		
	
	
			54 lines
		
	
	
		
			2.8 KiB
		
	
	
	
		
			C
		
	
| 
											10 years ago
										 | /*
 | ||
|  |  *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. | ||
|  |  * | ||
|  |  *  Use of this source code is governed by a BSD-style license | ||
|  |  *  that can be found in the LICENSE file in the root of the source | ||
|  |  *  tree. An additional intellectual property rights grant can be found | ||
|  |  *  in the file PATENTS.  All contributing project authors may | ||
|  |  *  be found in the AUTHORS file in the root of the source tree. | ||
|  |  */ | ||
|  | 
 | ||
|  | #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_DEFINES_H_
 | ||
|  | #define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_DEFINES_H_
 | ||
|  | 
 | ||
|  | //#define PROCESS_FLOW_0    // Use the traditional method.
 | ||
|  | //#define PROCESS_FLOW_1    // Use traditional with DD estimate of prior SNR.
 | ||
|  | #define PROCESS_FLOW_2    // Use the new method of speech/noise classification.
 | ||
|  | 
 | ||
|  | #define BLOCKL_MAX          160 // max processing block length: 160
 | ||
|  | #define ANAL_BLOCKL_MAX     256 // max analysis block length: 256
 | ||
|  | #define HALF_ANAL_BLOCKL    129 // half max analysis block length + 1
 | ||
|  | 
 | ||
|  | #define QUANTILE            (float)0.25
 | ||
|  | 
 | ||
|  | #define SIMULT              3
 | ||
|  | #define END_STARTUP_LONG    200
 | ||
|  | #define END_STARTUP_SHORT   50
 | ||
|  | #define FACTOR              (float)40.0
 | ||
|  | #define WIDTH               (float)0.01
 | ||
|  | 
 | ||
|  | #define SMOOTH              (float)0.75 // filter smoothing
 | ||
|  | // Length of fft work arrays.
 | ||
|  | #define IP_LENGTH (ANAL_BLOCKL_MAX >> 1) // must be at least ceil(2 + sqrt(ANAL_BLOCKL_MAX/2))
 | ||
|  | #define W_LENGTH (ANAL_BLOCKL_MAX >> 1)
 | ||
|  | 
 | ||
|  | //PARAMETERS FOR NEW METHOD
 | ||
|  | #define DD_PR_SNR           (float)0.98 // DD update of prior SNR
 | ||
|  | #define LRT_TAVG            (float)0.50 // tavg parameter for LRT (previously 0.90)
 | ||
|  | #define SPECT_FL_TAVG       (float)0.30 // tavg parameter for spectral flatness measure
 | ||
|  | #define SPECT_DIFF_TAVG     (float)0.30 // tavg parameter for spectral difference measure
 | ||
|  | #define PRIOR_UPDATE        (float)0.10 // update parameter of prior model
 | ||
|  | #define NOISE_UPDATE        (float)0.90 // update parameter for noise
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|  | #define SPEECH_UPDATE       (float)0.99 // update parameter when likely speech
 | ||
|  | #define WIDTH_PR_MAP        (float)4.0  // width parameter in sigmoid map for prior model
 | ||
|  | #define LRT_FEATURE_THR     (float)0.5  // default threshold for LRT feature
 | ||
|  | #define SF_FEATURE_THR      (float)0.5  // default threshold for Spectral Flatness feature
 | ||
|  | #define SD_FEATURE_THR      (float)0.5  // default threshold for Spectral Difference feature
 | ||
|  | #define PROB_RANGE          (float)0.20 // probability threshold for noise state in
 | ||
|  |                                         // speech/noise likelihood
 | ||
|  | #define HIST_PAR_EST         1000       // histogram size for estimation of parameters
 | ||
|  | #define GAMMA_PAUSE         (float)0.05 // update for conservative noise estimate
 | ||
|  | //
 | ||
|  | #define B_LIM               (float)0.5  // threshold in final energy gain factor calculation
 | ||
|  | #endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_DEFINES_H_
 |