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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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 |
11 #include "webrtc/modules/audio_coding/neteq/expand.h" | 11 #include "webrtc/modules/audio_coding/neteq/expand.h" |
12 | 12 |
13 #include <assert.h> | 13 #include <assert.h> |
14 #include <string.h> // memset | 14 #include <string.h> // memset |
15 | 15 |
16 #include <algorithm> // min, max | 16 #include <algorithm> // min, max |
17 #include <limits> // numeric_limits<T> | 17 #include <limits> // numeric_limits<T> |
18 | 18 |
19 #include "webrtc/base/safe_conversions.h" | 19 #include "webrtc/base/safe_conversions.h" |
20 #include "webrtc/common_audio/signal_processing/include/signal_processing_librar
y.h" | 20 #include "webrtc/common_audio/signal_processing/include/signal_processing_librar
y.h" |
21 #include "webrtc/modules/audio_coding/neteq/background_noise.h" | 21 #include "webrtc/modules/audio_coding/neteq/background_noise.h" |
| 22 #include "webrtc/modules/audio_coding/neteq/cross_correlation.h" |
22 #include "webrtc/modules/audio_coding/neteq/dsp_helper.h" | 23 #include "webrtc/modules/audio_coding/neteq/dsp_helper.h" |
23 #include "webrtc/modules/audio_coding/neteq/random_vector.h" | 24 #include "webrtc/modules/audio_coding/neteq/random_vector.h" |
24 #include "webrtc/modules/audio_coding/neteq/statistics_calculator.h" | 25 #include "webrtc/modules/audio_coding/neteq/statistics_calculator.h" |
25 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h" | 26 #include "webrtc/modules/audio_coding/neteq/sync_buffer.h" |
26 | 27 |
27 namespace webrtc { | 28 namespace webrtc { |
28 | 29 |
29 Expand::Expand(BackgroundNoise* background_noise, | 30 Expand::Expand(BackgroundNoise* background_noise, |
30 SyncBuffer* sync_buffer, | 31 SyncBuffer* sync_buffer, |
31 RandomVector* random_vector, | 32 RandomVector* random_vector, |
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372 size_t fs_mult_lpc_analysis_len = fs_mult * kLpcAnalysisLength; | 373 size_t fs_mult_lpc_analysis_len = fs_mult * kLpcAnalysisLength; |
373 | 374 |
374 const size_t signal_length = static_cast<size_t>(256 * fs_mult); | 375 const size_t signal_length = static_cast<size_t>(256 * fs_mult); |
375 const int16_t* audio_history = | 376 const int16_t* audio_history = |
376 &(*sync_buffer_)[0][sync_buffer_->Size() - signal_length]; | 377 &(*sync_buffer_)[0][sync_buffer_->Size() - signal_length]; |
377 | 378 |
378 // Initialize. | 379 // Initialize. |
379 InitializeForAnExpandPeriod(); | 380 InitializeForAnExpandPeriod(); |
380 | 381 |
381 // Calculate correlation in downsampled domain (4 kHz sample rate). | 382 // Calculate correlation in downsampled domain (4 kHz sample rate). |
382 int correlation_scale; | |
383 size_t correlation_length = 51; // TODO(hlundin): Legacy bit-exactness. | 383 size_t correlation_length = 51; // TODO(hlundin): Legacy bit-exactness. |
384 // If it is decided to break bit-exactness |correlation_length| should be | 384 // If it is decided to break bit-exactness |correlation_length| should be |
385 // initialized to the return value of Correlation(). | 385 // initialized to the return value of Correlation(). |
386 Correlation(audio_history, signal_length, correlation_vector, | 386 Correlation(audio_history, signal_length, correlation_vector); |
387 &correlation_scale); | |
388 | 387 |
389 // Find peaks in correlation vector. | 388 // Find peaks in correlation vector. |
390 DspHelper::PeakDetection(correlation_vector, correlation_length, | 389 DspHelper::PeakDetection(correlation_vector, correlation_length, |
391 kNumCorrelationCandidates, fs_mult, | 390 kNumCorrelationCandidates, fs_mult, |
392 best_correlation_index, best_correlation); | 391 best_correlation_index, best_correlation); |
393 | 392 |
394 // Adjust peak locations; cross-correlation lags start at 2.5 ms | 393 // Adjust peak locations; cross-correlation lags start at 2.5 ms |
395 // (20 * fs_mult samples). | 394 // (20 * fs_mult samples). |
396 best_correlation_index[0] += fs_mult_20; | 395 best_correlation_index[0] += fs_mult_20; |
397 best_correlation_index[1] += fs_mult_20; | 396 best_correlation_index[1] += fs_mult_20; |
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448 WEBRTC_SPL_ABS_W16((distortion_lag-correlation_lag)) + 1); | 447 WEBRTC_SPL_ABS_W16((distortion_lag-correlation_lag)) + 1); |
449 assert(correlation_lags <= static_cast<size_t>(99 * fs_mult + 1)); | 448 assert(correlation_lags <= static_cast<size_t>(99 * fs_mult + 1)); |
450 | 449 |
451 for (size_t channel_ix = 0; channel_ix < num_channels_; ++channel_ix) { | 450 for (size_t channel_ix = 0; channel_ix < num_channels_; ++channel_ix) { |
452 ChannelParameters& parameters = channel_parameters_[channel_ix]; | 451 ChannelParameters& parameters = channel_parameters_[channel_ix]; |
453 // Calculate suitable scaling. | 452 // Calculate suitable scaling. |
454 int16_t signal_max = WebRtcSpl_MaxAbsValueW16( | 453 int16_t signal_max = WebRtcSpl_MaxAbsValueW16( |
455 &audio_history[signal_length - correlation_length - start_index | 454 &audio_history[signal_length - correlation_length - start_index |
456 - correlation_lags], | 455 - correlation_lags], |
457 correlation_length + start_index + correlation_lags - 1); | 456 correlation_length + start_index + correlation_lags - 1); |
458 correlation_scale = (31 - WebRtcSpl_NormW32(signal_max * signal_max)) + | 457 int correlation_scale = (31 - WebRtcSpl_NormW32(signal_max * signal_max)) + |
459 (31 - WebRtcSpl_NormW32(static_cast<int32_t>(correlation_length))) - 31; | 458 (31 - WebRtcSpl_NormW32(static_cast<int32_t>(correlation_length))) - 31; |
460 correlation_scale = std::max(0, correlation_scale); | 459 correlation_scale = std::max(0, correlation_scale); |
461 | 460 |
462 // Calculate the correlation, store in |correlation_vector2|. | 461 // Calculate the correlation, store in |correlation_vector2|. |
463 WebRtcSpl_CrossCorrelation( | 462 WebRtcSpl_CrossCorrelation( |
464 correlation_vector2, | 463 correlation_vector2, |
465 &(audio_history[signal_length - correlation_length]), | 464 &(audio_history[signal_length - correlation_length]), |
466 &(audio_history[signal_length - correlation_length - start_index]), | 465 &(audio_history[signal_length - correlation_length - start_index]), |
467 correlation_length, correlation_lags, correlation_scale, -1); | 466 correlation_length, correlation_lags, correlation_scale, -1); |
468 | 467 |
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575 expand_lags_[1] = (distortion_lag + correlation_lag) / 2; | 574 expand_lags_[1] = (distortion_lag + correlation_lag) / 2; |
576 // Third lag is the average again, but rounding towards |correlation_lag|. | 575 // Third lag is the average again, but rounding towards |correlation_lag|. |
577 if (distortion_lag > correlation_lag) { | 576 if (distortion_lag > correlation_lag) { |
578 expand_lags_[2] = (distortion_lag + correlation_lag - 1) / 2; | 577 expand_lags_[2] = (distortion_lag + correlation_lag - 1) / 2; |
579 } else { | 578 } else { |
580 expand_lags_[2] = (distortion_lag + correlation_lag + 1) / 2; | 579 expand_lags_[2] = (distortion_lag + correlation_lag + 1) / 2; |
581 } | 580 } |
582 } | 581 } |
583 | 582 |
584 // Calculate the LPC and the gain of the filters. | 583 // Calculate the LPC and the gain of the filters. |
585 // Calculate scale value needed for auto-correlation. | |
586 correlation_scale = WebRtcSpl_MaxAbsValueW16( | |
587 &(audio_history[signal_length - fs_mult_lpc_analysis_len]), | |
588 fs_mult_lpc_analysis_len); | |
589 | |
590 correlation_scale = std::min(16 - WebRtcSpl_NormW32(correlation_scale), 0); | |
591 correlation_scale = std::max(correlation_scale * 2 + 7, 0); | |
592 | 584 |
593 // Calculate kUnvoicedLpcOrder + 1 lags of the auto-correlation function. | 585 // Calculate kUnvoicedLpcOrder + 1 lags of the auto-correlation function. |
594 size_t temp_index = signal_length - fs_mult_lpc_analysis_len - | 586 size_t temp_index = signal_length - fs_mult_lpc_analysis_len - |
595 kUnvoicedLpcOrder; | 587 kUnvoicedLpcOrder; |
596 // Copy signal to temporary vector to be able to pad with leading zeros. | 588 // Copy signal to temporary vector to be able to pad with leading zeros. |
597 int16_t* temp_signal = new int16_t[fs_mult_lpc_analysis_len | 589 int16_t* temp_signal = new int16_t[fs_mult_lpc_analysis_len |
598 + kUnvoicedLpcOrder]; | 590 + kUnvoicedLpcOrder]; |
599 memset(temp_signal, 0, | 591 memset(temp_signal, 0, |
600 sizeof(int16_t) * (fs_mult_lpc_analysis_len + kUnvoicedLpcOrder)); | 592 sizeof(int16_t) * (fs_mult_lpc_analysis_len + kUnvoicedLpcOrder)); |
601 memcpy(&temp_signal[kUnvoicedLpcOrder], | 593 memcpy(&temp_signal[kUnvoicedLpcOrder], |
602 &audio_history[temp_index + kUnvoicedLpcOrder], | 594 &audio_history[temp_index + kUnvoicedLpcOrder], |
603 sizeof(int16_t) * fs_mult_lpc_analysis_len); | 595 sizeof(int16_t) * fs_mult_lpc_analysis_len); |
604 WebRtcSpl_CrossCorrelation(auto_correlation, | 596 CrossCorrelationWithAutoShift( |
605 &temp_signal[kUnvoicedLpcOrder], | 597 &temp_signal[kUnvoicedLpcOrder], &temp_signal[kUnvoicedLpcOrder], |
606 &temp_signal[kUnvoicedLpcOrder], | 598 fs_mult_lpc_analysis_len, kUnvoicedLpcOrder + 1, -1, auto_correlation); |
607 fs_mult_lpc_analysis_len, kUnvoicedLpcOrder + 1, | |
608 correlation_scale, -1); | |
609 delete [] temp_signal; | 599 delete [] temp_signal; |
610 | 600 |
611 // Verify that variance is positive. | 601 // Verify that variance is positive. |
612 if (auto_correlation[0] > 0) { | 602 if (auto_correlation[0] > 0) { |
613 // Estimate AR filter parameters using Levinson-Durbin algorithm; | 603 // Estimate AR filter parameters using Levinson-Durbin algorithm; |
614 // kUnvoicedLpcOrder + 1 filter coefficients. | 604 // kUnvoicedLpcOrder + 1 filter coefficients. |
615 int16_t stability = WebRtcSpl_LevinsonDurbin(auto_correlation, | 605 int16_t stability = WebRtcSpl_LevinsonDurbin(auto_correlation, |
616 parameters.ar_filter, | 606 parameters.ar_filter, |
617 reflection_coeff, | 607 reflection_coeff, |
618 kUnvoicedLpcOrder); | 608 kUnvoicedLpcOrder); |
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759 voice_mix_factor(0), | 749 voice_mix_factor(0), |
760 current_voice_mix_factor(0), | 750 current_voice_mix_factor(0), |
761 onset(false), | 751 onset(false), |
762 mute_slope(0) { | 752 mute_slope(0) { |
763 memset(ar_filter, 0, sizeof(ar_filter)); | 753 memset(ar_filter, 0, sizeof(ar_filter)); |
764 memset(ar_filter_state, 0, sizeof(ar_filter_state)); | 754 memset(ar_filter_state, 0, sizeof(ar_filter_state)); |
765 } | 755 } |
766 | 756 |
767 void Expand::Correlation(const int16_t* input, | 757 void Expand::Correlation(const int16_t* input, |
768 size_t input_length, | 758 size_t input_length, |
769 int16_t* output, | 759 int16_t* output) const { |
770 int* output_scale) const { | |
771 // Set parameters depending on sample rate. | 760 // Set parameters depending on sample rate. |
772 const int16_t* filter_coefficients; | 761 const int16_t* filter_coefficients; |
773 size_t num_coefficients; | 762 size_t num_coefficients; |
774 int16_t downsampling_factor; | 763 int16_t downsampling_factor; |
775 if (fs_hz_ == 8000) { | 764 if (fs_hz_ == 8000) { |
776 num_coefficients = 3; | 765 num_coefficients = 3; |
777 downsampling_factor = 2; | 766 downsampling_factor = 2; |
778 filter_coefficients = DspHelper::kDownsample8kHzTbl; | 767 filter_coefficients = DspHelper::kDownsample8kHzTbl; |
779 } else if (fs_hz_ == 16000) { | 768 } else if (fs_hz_ == 16000) { |
780 num_coefficients = 5; | 769 num_coefficients = 5; |
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807 downsampling_factor, kFilterDelay); | 796 downsampling_factor, kFilterDelay); |
808 | 797 |
809 // Normalize |downsampled_input| to using all 16 bits. | 798 // Normalize |downsampled_input| to using all 16 bits. |
810 int16_t max_value = WebRtcSpl_MaxAbsValueW16(downsampled_input, | 799 int16_t max_value = WebRtcSpl_MaxAbsValueW16(downsampled_input, |
811 kDownsampledLength); | 800 kDownsampledLength); |
812 int16_t norm_shift = 16 - WebRtcSpl_NormW32(max_value); | 801 int16_t norm_shift = 16 - WebRtcSpl_NormW32(max_value); |
813 WebRtcSpl_VectorBitShiftW16(downsampled_input, kDownsampledLength, | 802 WebRtcSpl_VectorBitShiftW16(downsampled_input, kDownsampledLength, |
814 downsampled_input, norm_shift); | 803 downsampled_input, norm_shift); |
815 | 804 |
816 int32_t correlation[kNumCorrelationLags]; | 805 int32_t correlation[kNumCorrelationLags]; |
817 static const int kCorrelationShift = 6; | 806 CrossCorrelationWithAutoShift( |
818 WebRtcSpl_CrossCorrelation( | |
819 correlation, | |
820 &downsampled_input[kDownsampledLength - kCorrelationLength], | 807 &downsampled_input[kDownsampledLength - kCorrelationLength], |
821 &downsampled_input[kDownsampledLength - kCorrelationLength | 808 &downsampled_input[kDownsampledLength - kCorrelationLength |
822 - kCorrelationStartLag], | 809 - kCorrelationStartLag], |
823 kCorrelationLength, kNumCorrelationLags, kCorrelationShift, -1); | 810 kCorrelationLength, kNumCorrelationLags, -1, correlation); |
824 | 811 |
825 // Normalize and move data from 32-bit to 16-bit vector. | 812 // Normalize and move data from 32-bit to 16-bit vector. |
826 int32_t max_correlation = WebRtcSpl_MaxAbsValueW32(correlation, | 813 int32_t max_correlation = WebRtcSpl_MaxAbsValueW32(correlation, |
827 kNumCorrelationLags); | 814 kNumCorrelationLags); |
828 int16_t norm_shift2 = static_cast<int16_t>( | 815 int16_t norm_shift2 = static_cast<int16_t>( |
829 std::max(18 - WebRtcSpl_NormW32(max_correlation), 0)); | 816 std::max(18 - WebRtcSpl_NormW32(max_correlation), 0)); |
830 WebRtcSpl_VectorBitShiftW32ToW16(output, kNumCorrelationLags, correlation, | 817 WebRtcSpl_VectorBitShiftW32ToW16(output, kNumCorrelationLags, correlation, |
831 norm_shift2); | 818 norm_shift2); |
832 // Total scale factor (right shifts) of correlation value. | |
833 *output_scale = 2 * norm_shift + kCorrelationShift + norm_shift2; | |
834 } | 819 } |
835 | 820 |
836 void Expand::UpdateLagIndex() { | 821 void Expand::UpdateLagIndex() { |
837 current_lag_index_ = current_lag_index_ + lag_index_direction_; | 822 current_lag_index_ = current_lag_index_ + lag_index_direction_; |
838 // Change direction if needed. | 823 // Change direction if needed. |
839 if (current_lag_index_ <= 0) { | 824 if (current_lag_index_ <= 0) { |
840 lag_index_direction_ = 1; | 825 lag_index_direction_ = 1; |
841 } | 826 } |
842 if (current_lag_index_ >= kNumLags - 1) { | 827 if (current_lag_index_ >= kNumLags - 1) { |
843 lag_index_direction_ = -1; | 828 lag_index_direction_ = -1; |
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953 const size_t kMaxRandSamples = RandomVector::kRandomTableSize; | 938 const size_t kMaxRandSamples = RandomVector::kRandomTableSize; |
954 while (samples_generated < length) { | 939 while (samples_generated < length) { |
955 size_t rand_length = std::min(length - samples_generated, kMaxRandSamples); | 940 size_t rand_length = std::min(length - samples_generated, kMaxRandSamples); |
956 random_vector_->IncreaseSeedIncrement(seed_increment); | 941 random_vector_->IncreaseSeedIncrement(seed_increment); |
957 random_vector_->Generate(rand_length, &random_vector[samples_generated]); | 942 random_vector_->Generate(rand_length, &random_vector[samples_generated]); |
958 samples_generated += rand_length; | 943 samples_generated += rand_length; |
959 } | 944 } |
960 } | 945 } |
961 | 946 |
962 } // namespace webrtc | 947 } // namespace webrtc |
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