/* * Copyright (c) 2012 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. */ #include "webrtc/modules/audio_coding/neteq/time_stretch.h" #include // min, max #include "webrtc/base/safe_conversions.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" #include "webrtc/modules/audio_coding/neteq/background_noise.h" #include "webrtc/modules/audio_coding/neteq/dsp_helper.h" namespace webrtc { TimeStretch::ReturnCodes TimeStretch::Process(const int16_t* input, size_t input_len, bool fast_mode, AudioMultiVector* output, size_t* length_change_samples) { // Pre-calculate common multiplication with |fs_mult_|. size_t fs_mult_120 = static_cast(fs_mult_ * 120); // Corresponds to 15 ms. const int16_t* signal; rtc::scoped_ptr signal_array; size_t signal_len; if (num_channels_ == 1) { signal = input; signal_len = input_len; } else { // We want |signal| to be only the first channel of |input|, which is // interleaved. Thus, we take the first sample, skip forward |num_channels| // samples, and continue like that. signal_len = input_len / num_channels_; signal_array.reset(new int16_t[signal_len]); signal = signal_array.get(); size_t j = master_channel_; for (size_t i = 0; i < signal_len; ++i) { signal_array[i] = input[j]; j += num_channels_; } } // Find maximum absolute value of input signal. max_input_value_ = WebRtcSpl_MaxAbsValueW16(signal, signal_len); // Downsample to 4 kHz sample rate and calculate auto-correlation. DspHelper::DownsampleTo4kHz(signal, signal_len, kDownsampledLen, sample_rate_hz_, true /* compensate delay*/, downsampled_input_); AutoCorrelation(); // Find the strongest correlation peak. static const size_t kNumPeaks = 1; size_t peak_index; int16_t peak_value; DspHelper::PeakDetection(auto_correlation_, kCorrelationLen, kNumPeaks, fs_mult_, &peak_index, &peak_value); // Assert that |peak_index| stays within boundaries. assert(peak_index <= (2 * kCorrelationLen - 1) * fs_mult_); // Compensate peak_index for displaced starting position. The displacement // happens in AutoCorrelation(). Here, |kMinLag| is in the down-sampled 4 kHz // domain, while the |peak_index| is in the original sample rate; hence, the // multiplication by fs_mult_ * 2. peak_index += kMinLag * fs_mult_ * 2; // Assert that |peak_index| stays within boundaries. assert(peak_index >= static_cast(20 * fs_mult_)); assert(peak_index <= 20 * fs_mult_ + (2 * kCorrelationLen - 1) * fs_mult_); // Calculate scaling to ensure that |peak_index| samples can be square-summed // without overflowing. int scaling = 31 - WebRtcSpl_NormW32(max_input_value_ * max_input_value_) - WebRtcSpl_NormW32(static_cast(peak_index)); scaling = std::max(0, scaling); // |vec1| starts at 15 ms minus one pitch period. const int16_t* vec1 = &signal[fs_mult_120 - peak_index]; // |vec2| start at 15 ms. const int16_t* vec2 = &signal[fs_mult_120]; // Calculate energies for |vec1| and |vec2|, assuming they both contain // |peak_index| samples. int32_t vec1_energy = WebRtcSpl_DotProductWithScale(vec1, vec1, peak_index, scaling); int32_t vec2_energy = WebRtcSpl_DotProductWithScale(vec2, vec2, peak_index, scaling); // Calculate cross-correlation between |vec1| and |vec2|. int32_t cross_corr = WebRtcSpl_DotProductWithScale(vec1, vec2, peak_index, scaling); // Check if the signal seems to be active speech or not (simple VAD). bool active_speech = SpeechDetection(vec1_energy, vec2_energy, peak_index, scaling); int16_t best_correlation; if (!active_speech) { SetParametersForPassiveSpeech(signal_len, &best_correlation, &peak_index); } else { // Calculate correlation: // cross_corr / sqrt(vec1_energy * vec2_energy). // Start with calculating scale values. int energy1_scale = std::max(0, 16 - WebRtcSpl_NormW32(vec1_energy)); int energy2_scale = std::max(0, 16 - WebRtcSpl_NormW32(vec2_energy)); // Make sure total scaling is even (to simplify scale factor after sqrt). if ((energy1_scale + energy2_scale) & 1) { // The sum is odd. energy1_scale += 1; } // Scale energies to int16_t. int16_t vec1_energy_int16 = static_cast(vec1_energy >> energy1_scale); int16_t vec2_energy_int16 = static_cast(vec2_energy >> energy2_scale); // Calculate square-root of energy product. int16_t sqrt_energy_prod = WebRtcSpl_SqrtFloor(vec1_energy_int16 * vec2_energy_int16); // Calculate cross_corr / sqrt(en1*en2) in Q14. int temp_scale = 14 - (energy1_scale + energy2_scale) / 2; cross_corr = WEBRTC_SPL_SHIFT_W32(cross_corr, temp_scale); cross_corr = std::max(0, cross_corr); // Don't use if negative. best_correlation = WebRtcSpl_DivW32W16(cross_corr, sqrt_energy_prod); // Make sure |best_correlation| is no larger than 1 in Q14. best_correlation = std::min(static_cast(16384), best_correlation); } // Check accelerate criteria and stretch the signal. ReturnCodes return_value = CheckCriteriaAndStretch(input, input_len, peak_index, best_correlation, active_speech, fast_mode, output); switch (return_value) { case kSuccess: *length_change_samples = peak_index; break; case kSuccessLowEnergy: *length_change_samples = peak_index; break; case kNoStretch: case kError: *length_change_samples = 0; break; } return return_value; } void TimeStretch::AutoCorrelation() { // Set scaling factor for cross correlation to protect against overflow. int scaling = kLogCorrelationLen - WebRtcSpl_NormW32( max_input_value_ * max_input_value_); scaling = std::max(0, scaling); // Calculate correlation from lag kMinLag to lag kMaxLag in 4 kHz domain. int32_t auto_corr[kCorrelationLen]; WebRtcSpl_CrossCorrelation(auto_corr, &downsampled_input_[kMaxLag], &downsampled_input_[kMaxLag - kMinLag], kCorrelationLen, kMaxLag - kMinLag, scaling, -1); // Normalize correlation to 14 bits and write to |auto_correlation_|. int32_t max_corr = WebRtcSpl_MaxAbsValueW32(auto_corr, kCorrelationLen); scaling = std::max(0, 17 - WebRtcSpl_NormW32(max_corr)); WebRtcSpl_VectorBitShiftW32ToW16(auto_correlation_, kCorrelationLen, auto_corr, scaling); } bool TimeStretch::SpeechDetection(int32_t vec1_energy, int32_t vec2_energy, size_t peak_index, int scaling) const { // Check if the signal seems to be active speech or not (simple VAD). // If (vec1_energy + vec2_energy) / (2 * peak_index) <= // 8 * background_noise_energy, then we say that the signal contains no // active speech. // Rewrite the inequality as: // (vec1_energy + vec2_energy) / 16 <= peak_index * background_noise_energy. // The two sides of the inequality will be denoted |left_side| and // |right_side|. int32_t left_side = (vec1_energy + vec2_energy) / 16; int32_t right_side; if (background_noise_.initialized()) { right_side = background_noise_.Energy(master_channel_); } else { // If noise parameters have not been estimated, use a fixed threshold. right_side = 75000; } int right_scale = 16 - WebRtcSpl_NormW32(right_side); right_scale = std::max(0, right_scale); left_side = left_side >> right_scale; right_side = rtc::checked_cast(peak_index) * (right_side >> right_scale); // Scale |left_side| properly before comparing with |right_side|. // (|scaling| is the scale factor before energy calculation, thus the scale // factor for the energy is 2 * scaling.) if (WebRtcSpl_NormW32(left_side) < 2 * scaling) { // Cannot scale only |left_side|, must scale |right_side| too. int temp_scale = WebRtcSpl_NormW32(left_side); left_side = left_side << temp_scale; right_side = right_side >> (2 * scaling - temp_scale); } else { left_side = left_side << 2 * scaling; } return left_side > right_side; } } // namespace webrtc