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			89 lines
		
	
	
		
			3.4 KiB
		
	
	
	
		
			C++
		
	
			
		
		
	
	
			89 lines
		
	
	
		
			3.4 KiB
		
	
	
	
		
			C++
		
	
| /*
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|  *  Copyright (c) 2012 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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| #include "webrtc/modules/audio_coding/neteq/accelerate.h"
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| 
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| #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
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| 
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| namespace webrtc {
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| 
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| Accelerate::ReturnCodes Accelerate::Process(
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|     const int16_t* input,
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|     size_t input_length,
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|     AudioMultiVector* output,
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|     int16_t* length_change_samples) {
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|   // Input length must be (almost) 30 ms.
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|   static const int k15ms = 120;  // 15 ms = 120 samples at 8 kHz sample rate.
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|   if (num_channels_ == 0 || static_cast<int>(input_length) / num_channels_ <
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|       (2 * k15ms - 1) * fs_mult_) {
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|     // Length of input data too short to do accelerate. Simply move all data
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|     // from input to output.
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|     output->PushBackInterleaved(input, input_length);
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|     return kError;
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|   }
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|   return TimeStretch::Process(input, input_length, output,
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|                               length_change_samples);
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| }
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| 
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| void Accelerate::SetParametersForPassiveSpeech(size_t /*len*/,
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|                                                int16_t* best_correlation,
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|                                                int* /*peak_index*/) const {
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|   // When the signal does not contain any active speech, the correlation does
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|   // not matter. Simply set it to zero.
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|   *best_correlation = 0;
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| }
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| 
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| Accelerate::ReturnCodes Accelerate::CheckCriteriaAndStretch(
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|     const int16_t* input, size_t input_length, size_t peak_index,
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|     int16_t best_correlation, bool active_speech,
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|     AudioMultiVector* output) const {
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|   // Check for strong correlation or passive speech.
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|   if ((best_correlation > kCorrelationThreshold) || !active_speech) {
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|     // Do accelerate operation by overlap add.
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| 
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|     // Pre-calculate common multiplication with |fs_mult_|.
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|     // 120 corresponds to 15 ms.
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|     size_t fs_mult_120 = fs_mult_ * 120;
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| 
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|     assert(fs_mult_120 >= peak_index);  // Should be handled in Process().
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|     // Copy first part; 0 to 15 ms.
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|     output->PushBackInterleaved(input, fs_mult_120 * num_channels_);
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|     // Copy the |peak_index| starting at 15 ms to |temp_vector|.
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|     AudioMultiVector temp_vector(num_channels_);
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|     temp_vector.PushBackInterleaved(&input[fs_mult_120 * num_channels_],
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|                                     peak_index * num_channels_);
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|     // Cross-fade |temp_vector| onto the end of |output|.
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|     output->CrossFade(temp_vector, peak_index);
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|     // Copy the last unmodified part, 15 ms + pitch period until the end.
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|     output->PushBackInterleaved(
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|         &input[(fs_mult_120 + peak_index) * num_channels_],
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|         input_length - (fs_mult_120 + peak_index) * num_channels_);
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| 
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|     if (active_speech) {
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|       return kSuccess;
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|     } else {
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|       return kSuccessLowEnergy;
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|     }
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|   } else {
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|     // Accelerate not allowed. Simply move all data from decoded to outData.
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|     output->PushBackInterleaved(input, input_length);
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|     return kNoStretch;
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|   }
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| }
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| 
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| Accelerate* AccelerateFactory::Create(
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|     int sample_rate_hz,
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|     size_t num_channels,
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|     const BackgroundNoise& background_noise) const {
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|   return new Accelerate(sample_rate_hz, num_channels, background_noise);
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| }
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| 
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| }  // namespace webrtc
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