Initial upload of NetEq4
This is the first public upload of the new NetEq, version 4.
It has been through extensive internal review during the course of
the project.
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/1073005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3425 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq4/expand.cc b/webrtc/modules/audio_coding/neteq4/expand.cc
new file mode 100644
index 0000000..6ea3203
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq4/expand.cc
@@ -0,0 +1,860 @@
+/*
+ * 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/neteq4/expand.h"
+
+#include <assert.h>
+
+#include <algorithm> // min, max
+#include <cstring> // memset
+
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/modules/audio_coding/neteq4/background_noise.h"
+#include "webrtc/modules/audio_coding/neteq4/dsp_helper.h"
+#include "webrtc/modules/audio_coding/neteq4/random_vector.h"
+#include "webrtc/modules/audio_coding/neteq4/sync_buffer.h"
+
+namespace webrtc {
+
+void Expand::Reset() {
+ first_expand_ = true;
+ consecutive_expands_ = 0;
+ max_lag_ = 0;
+ for (size_t ix = 0; ix < num_channels_; ++ix) {
+ channel_parameters_[ix].expand_vector0.Clear();
+ channel_parameters_[ix].expand_vector1.Clear();
+ }
+}
+
+int Expand::Process(AudioMultiVector<int16_t>* output) {
+ int16_t random_vector[kMaxSampleRate / 8000 * 120 + 30];
+ int16_t scaled_random_vector[kMaxSampleRate / 8000 * 125];
+ static const int kTempDataSize = 3600;
+ int16_t temp_data[kTempDataSize]; // TODO(hlundin) Remove this.
+ int16_t* voiced_vector_storage = temp_data;
+ int16_t* voiced_vector = &voiced_vector_storage[overlap_length_];
+ static const int kNoiseLpcOrder = BackgroundNoise::kMaxLpcOrder;
+ int16_t unvoiced_array_memory[kNoiseLpcOrder + kMaxSampleRate / 8000 * 125];
+ int16_t* unvoiced_vector = unvoiced_array_memory + kUnvoicedLpcOrder;
+ int16_t* noise_vector = unvoiced_array_memory + kNoiseLpcOrder;
+
+ int fs_mult = fs_hz_ / 8000;
+
+ if (first_expand_) {
+ // Perform initial setup if this is the first expansion since last reset.
+ AnalyzeSignal(random_vector);
+ first_expand_ = false;
+ } else {
+ // This is not the first expansion, parameters are already estimated.
+ // Extract a noise segment.
+ int16_t rand_length = max_lag_;
+ // TODO(hlundin): This if-statement should not be needed. Should be just
+ // as good to generate all of the vector in one call in either case.
+ if (rand_length <= RandomVector::kRandomTableSize) {
+ random_vector_->IncreaseSeedIncrement(2);
+ random_vector_->Generate(rand_length, random_vector);
+ } else {
+ // This only applies to SWB where length could be larger than 256.
+ assert(rand_length <= kMaxSampleRate / 8000 * 120 + 30);
+ random_vector_->IncreaseSeedIncrement(2);
+ random_vector_->Generate(RandomVector::kRandomTableSize, random_vector);
+ random_vector_->IncreaseSeedIncrement(2);
+ random_vector_->Generate(rand_length - RandomVector::kRandomTableSize,
+ &random_vector[RandomVector::kRandomTableSize]);
+ }
+ }
+
+
+ // Generate signal.
+ UpdateLagIndex();
+
+ // Voiced part.
+ // Generate a weighted vector with the current lag.
+ size_t expansion_vector_length = max_lag_ + overlap_length_;
+ size_t current_lag = expand_lags_[current_lag_index_];
+ // Copy lag+overlap data.
+ size_t expansion_vector_position = expansion_vector_length - current_lag -
+ overlap_length_;
+ size_t temp_length = current_lag + overlap_length_;
+ for (size_t channel_ix = 0; channel_ix < num_channels_; ++channel_ix) {
+ ChannelParameters& parameters = channel_parameters_[channel_ix];
+ if (current_lag_index_ == 0) {
+ // Use only expand_vector0.
+ assert(expansion_vector_position + temp_length <=
+ parameters.expand_vector0.Size());
+ memcpy(voiced_vector_storage,
+ ¶meters.expand_vector0[expansion_vector_position],
+ sizeof(int16_t) * temp_length);
+ } else if (current_lag_index_ == 1) {
+ // Mix 3/4 of expand_vector0 with 1/4 of expand_vector1.
+ WebRtcSpl_ScaleAndAddVectorsWithRound(
+ ¶meters.expand_vector0[expansion_vector_position], 3,
+ ¶meters.expand_vector1[expansion_vector_position], 1, 2,
+ voiced_vector_storage, temp_length);
+ } else if (current_lag_index_ == 2) {
+ // Mix 1/2 of expand_vector0 with 1/2 of expand_vector1.
+ assert(expansion_vector_position + temp_length <=
+ parameters.expand_vector0.Size());
+ assert(expansion_vector_position + temp_length <=
+ parameters.expand_vector1.Size());
+ WebRtcSpl_ScaleAndAddVectorsWithRound(
+ ¶meters.expand_vector0[expansion_vector_position], 1,
+ ¶meters.expand_vector1[expansion_vector_position], 1, 1,
+ voiced_vector_storage, temp_length);
+ }
+
+ // Get tapering window parameters. Values are in Q15.
+ int16_t muting_window, muting_window_increment;
+ int16_t unmuting_window, unmuting_window_increment;
+ if (fs_hz_ == 8000) {
+ muting_window = DspHelper::kMuteFactorStart8kHz;
+ muting_window_increment = DspHelper::kMuteFactorIncrement8kHz;
+ unmuting_window = DspHelper::kUnmuteFactorStart8kHz;
+ unmuting_window_increment = DspHelper::kUnmuteFactorIncrement8kHz;
+ } else if (fs_hz_ == 16000) {
+ muting_window = DspHelper::kMuteFactorStart16kHz;
+ muting_window_increment = DspHelper::kMuteFactorIncrement16kHz;
+ unmuting_window = DspHelper::kUnmuteFactorStart16kHz;
+ unmuting_window_increment = DspHelper::kUnmuteFactorIncrement16kHz;
+ } else if (fs_hz_ == 32000) {
+ muting_window = DspHelper::kMuteFactorStart32kHz;
+ muting_window_increment = DspHelper::kMuteFactorIncrement32kHz;
+ unmuting_window = DspHelper::kUnmuteFactorStart32kHz;
+ unmuting_window_increment = DspHelper::kUnmuteFactorIncrement32kHz;
+ } else { // fs_ == 48000
+ muting_window = DspHelper::kMuteFactorStart48kHz;
+ muting_window_increment = DspHelper::kMuteFactorIncrement48kHz;
+ unmuting_window = DspHelper::kUnmuteFactorStart48kHz;
+ unmuting_window_increment = DspHelper::kUnmuteFactorIncrement48kHz;
+ }
+
+ // Smooth the expanded if it has not been muted to a low amplitude and
+ // |current_voice_mix_factor| is larger than 0.5.
+ if ((parameters.mute_factor > 819) &&
+ (parameters.current_voice_mix_factor > 8192)) {
+ size_t start_ix = sync_buffer_->Size() - overlap_length_;
+ for (size_t i = 0; i < overlap_length_; i++) {
+ // Do overlap add between new vector and overlap.
+ (*sync_buffer_)[channel_ix][start_ix + i] =
+ (((*sync_buffer_)[channel_ix][start_ix + i] * muting_window) +
+ (((parameters.mute_factor * voiced_vector_storage[i]) >> 14) *
+ unmuting_window) + 16384) >> 15;
+ muting_window += muting_window_increment;
+ unmuting_window += unmuting_window_increment;
+ }
+ } else if (parameters.mute_factor == 0) {
+ // The expanded signal will consist of only comfort noise if
+ // mute_factor = 0. Set the output length to 15 ms for best noise
+ // production.
+ // TODO(hlundin): This has been disabled since the length of
+ // parameters.expand_vector0 and parameters.expand_vector1 no longer
+ // match with expand_lags_, causing invalid reads and writes. Is it a good
+ // idea to enable this again, and solve the vector size problem?
+// max_lag_ = fs_mult * 120;
+// expand_lags_[0] = fs_mult * 120;
+// expand_lags_[1] = fs_mult * 120;
+// expand_lags_[2] = fs_mult * 120;
+ }
+
+ // Unvoiced part.
+ // Filter |scaled_random_vector| through |ar_filter_|.
+ memcpy(unvoiced_vector - kUnvoicedLpcOrder, parameters.ar_filter_state,
+ sizeof(int16_t) * kUnvoicedLpcOrder);
+ int32_t add_constant = 0;
+ if (parameters.ar_gain_scale > 0) {
+ add_constant = 1 << (parameters.ar_gain_scale - 1);
+ }
+ WebRtcSpl_AffineTransformVector(scaled_random_vector, random_vector,
+ parameters.ar_gain, add_constant,
+ parameters.ar_gain_scale, current_lag);
+ WebRtcSpl_FilterARFastQ12(scaled_random_vector, unvoiced_vector,
+ parameters.ar_filter,
+ kUnvoicedLpcOrder + 1, current_lag);
+ memcpy(parameters.ar_filter_state,
+ &(unvoiced_vector[current_lag - kUnvoicedLpcOrder]),
+ sizeof(int16_t) * kUnvoicedLpcOrder);
+
+ // Combine voiced and unvoiced contributions.
+
+ // Set a suitable cross-fading slope.
+ // For lag =
+ // <= 31 * fs_mult => go from 1 to 0 in about 8 ms;
+ // (>= 31 .. <= 63) * fs_mult => go from 1 to 0 in about 16 ms;
+ // >= 64 * fs_mult => go from 1 to 0 in about 32 ms.
+ // temp_shift = getbits(max_lag_) - 5.
+ int temp_shift = (31 - WebRtcSpl_NormW32(max_lag_)) - 5;
+ int16_t mix_factor_increment = 256 >> temp_shift;
+ if (stop_muting_) {
+ mix_factor_increment = 0;
+ }
+
+ // Create combined signal by shifting in more and more of unvoiced part.
+ temp_shift = 8 - temp_shift; // = getbits(mix_factor_increment).
+ size_t temp_lenght = (parameters.current_voice_mix_factor -
+ parameters.voice_mix_factor) >> temp_shift;
+ temp_lenght = std::min(temp_lenght, current_lag);
+ DspHelper::CrossFade(voiced_vector, unvoiced_vector, temp_lenght,
+ ¶meters.current_voice_mix_factor,
+ mix_factor_increment, temp_data);
+
+ // End of cross-fading period was reached before end of expanded signal
+ // path. Mix the rest with a fixed mixing factor.
+ if (temp_lenght < current_lag) {
+ if (mix_factor_increment != 0) {
+ parameters.current_voice_mix_factor = parameters.voice_mix_factor;
+ }
+ int temp_scale = 16384 - parameters.current_voice_mix_factor;
+ WebRtcSpl_ScaleAndAddVectorsWithRound(
+ voiced_vector + temp_lenght, parameters.current_voice_mix_factor,
+ unvoiced_vector + temp_lenght, temp_scale, 14,
+ temp_data + temp_lenght, current_lag - temp_lenght);
+ }
+
+ // Select muting slope depending on how many consecutive expands we have
+ // done.
+ if (consecutive_expands_ == 3) {
+ // Let the mute factor decrease from 1.0 to 0.95 in 6.25 ms.
+ // mute_slope = 0.0010 / fs_mult in Q20.
+ parameters.mute_slope = std::max(parameters.mute_slope,
+ static_cast<int16_t>(1049 / fs_mult));
+ }
+ if (consecutive_expands_ == 7) {
+ // Let the mute factor decrease from 1.0 to 0.90 in 6.25 ms.
+ // mute_slope = 0.0020 / fs_mult in Q20.
+ parameters.mute_slope = std::max(parameters.mute_slope,
+ static_cast<int16_t>(2097 / fs_mult));
+ }
+
+ // Mute segment according to slope value.
+ if ((consecutive_expands_ != 0) || !parameters.onset) {
+ // Mute to the previous level, then continue with the muting.
+ WebRtcSpl_AffineTransformVector(temp_data, temp_data,
+ parameters.mute_factor, 8192,
+ 14, current_lag);
+
+ if (!stop_muting_) {
+ DspHelper::MuteSignal(temp_data, parameters.mute_slope, current_lag);
+
+ // Shift by 6 to go from Q20 to Q14.
+ // TODO(hlundin): Adding 8192 before shifting 6 steps seems wrong.
+ // Legacy.
+ int16_t gain = 16384 -
+ (((current_lag * parameters.mute_slope) + 8192) >> 6);
+ gain = ((gain * parameters.mute_factor) + 8192) >> 14;
+
+ // Guard against getting stuck with very small (but sometimes audible)
+ // gain.
+ if ((consecutive_expands_ > 3) && (gain >= parameters.mute_factor)) {
+ parameters.mute_factor = 0;
+ } else {
+ parameters.mute_factor = gain;
+ }
+ }
+ }
+
+ // Background noise part.
+ // TODO(hlundin): Move to separate method? In BackgroundNoise class?
+ if (background_noise_->initialized()) {
+ // Use background noise parameters.
+ memcpy(noise_vector - kNoiseLpcOrder,
+ background_noise_->FilterState(channel_ix),
+ sizeof(int16_t) * kNoiseLpcOrder);
+
+ if (background_noise_->ScaleShift(channel_ix) > 1) {
+ add_constant = 1 << (background_noise_->ScaleShift(channel_ix) - 1);
+ } else {
+ add_constant = 0;
+ }
+
+ // Scale random vector to correct energy level.
+ WebRtcSpl_AffineTransformVector(
+ scaled_random_vector, random_vector,
+ background_noise_->Scale(channel_ix), add_constant,
+ background_noise_->ScaleShift(channel_ix), current_lag);
+
+ WebRtcSpl_FilterARFastQ12(scaled_random_vector, noise_vector,
+ background_noise_->Filter(channel_ix),
+ kNoiseLpcOrder + 1,
+ current_lag);
+
+ background_noise_->SetFilterState(
+ channel_ix,
+ &(noise_vector[current_lag - kNoiseLpcOrder]),
+ kNoiseLpcOrder);
+
+ // Unmute the background noise.
+ int16_t bgn_mute_factor = background_noise_->MuteFactor(channel_ix);
+ BackgroundNoise::BackgroundNoiseMode bgn_mode = background_noise_->mode();
+ if (bgn_mode == BackgroundNoise::kBgnFade &&
+ consecutive_expands_ >= kMaxConsecutiveExpands &&
+ bgn_mute_factor > 0) {
+ // Fade BGN to zero.
+ // Calculate muting slope, approximately -2^18 / fs_hz.
+ int16_t mute_slope;
+ if (fs_hz_ == 8000) {
+ mute_slope = -32;
+ } else if (fs_hz_ == 16000) {
+ mute_slope = -16;
+ } else if (fs_hz_ == 32000) {
+ mute_slope = -8;
+ } else {
+ mute_slope = -5;
+ }
+ // Use UnmuteSignal function with negative slope.
+ // |bgn_mute_factor| is in Q14. |mute_slope| is in Q20.
+ DspHelper::UnmuteSignal(noise_vector, current_lag, &bgn_mute_factor,
+ mute_slope, noise_vector);
+ } else if (bgn_mute_factor < 16384) {
+ // If mode is kBgnOff, or if kBgnFade has started fading,
+ // Use regular |mute_slope|.
+ if (!stop_muting_ && bgn_mode != BackgroundNoise::kBgnOff &&
+ !(bgn_mode == BackgroundNoise::kBgnFade &&
+ consecutive_expands_ >= kMaxConsecutiveExpands)) {
+ DspHelper::UnmuteSignal(noise_vector, current_lag, &bgn_mute_factor,
+ parameters.mute_slope, noise_vector);
+ } else {
+ // kBgnOn and stop muting, or
+ // kBgnOff (mute factor is always 0), or
+ // kBgnFade has reached 0.
+ WebRtcSpl_AffineTransformVector(noise_vector, noise_vector,
+ bgn_mute_factor, 8192, 14,
+ current_lag);
+ }
+ }
+ // Update mute_factor in BackgroundNoise class.
+ background_noise_->SetMuteFactor(channel_ix, bgn_mute_factor);
+ } else {
+ // BGN parameters have not been initialized; use zero noise.
+ memset(noise_vector, 0, sizeof(int16_t) * current_lag);
+ }
+
+ // Add background noise to the combined voiced-unvoiced signal.
+ for (size_t i = 0; i < current_lag; i++) {
+ temp_data[i] = temp_data[i] + noise_vector[i];
+ }
+ if (channel_ix == 0) {
+ output->AssertSize(current_lag);
+ } else {
+ assert(output->Size() == current_lag);
+ }
+ memcpy(&(*output)[channel_ix][0], temp_data,
+ sizeof(temp_data[0]) * current_lag);
+ }
+
+ // Increase call number and cap it.
+ ++consecutive_expands_;
+ if (consecutive_expands_ > kMaxConsecutiveExpands) {
+ consecutive_expands_ = kMaxConsecutiveExpands;
+ }
+
+ return 0;
+}
+
+void Expand::SetParametersForNormalAfterExpand() {
+ current_lag_index_ = 0;
+ lag_index_direction_ = 0;
+ stop_muting_ = true; // Do not mute signal any more.
+}
+
+void Expand::SetParametersForMergeAfterExpand() {
+ current_lag_index_ = -1; /* out of the 3 possible ones */
+ lag_index_direction_ = 1; /* make sure we get the "optimal" lag */
+ stop_muting_ = true;
+}
+
+void Expand::AnalyzeSignal(int16_t* random_vector) {
+ int32_t auto_correlation[kUnvoicedLpcOrder + 1];
+ int16_t reflection_coeff[kUnvoicedLpcOrder];
+ int16_t correlation_vector[kMaxSampleRate / 8000 * 102];
+ int best_correlation_index[kNumCorrelationCandidates];
+ int16_t best_correlation[kNumCorrelationCandidates];
+ int16_t best_distortion_index[kNumCorrelationCandidates];
+ int16_t best_distortion[kNumCorrelationCandidates];
+ int32_t correlation_vector2[(99 * kMaxSampleRate / 8000) + 1];
+ int32_t best_distortion_w32[kNumCorrelationCandidates];
+ static const int kNoiseLpcOrder = BackgroundNoise::kMaxLpcOrder;
+ int16_t unvoiced_array_memory[kNoiseLpcOrder + kMaxSampleRate / 8000 * 125];
+ int16_t* unvoiced_vector = unvoiced_array_memory + kUnvoicedLpcOrder;
+
+ int fs_mult = fs_hz_ / 8000;
+
+ // Pre-calculate common multiplications with fs_mult.
+ int fs_mult_4 = fs_mult * 4;
+ int fs_mult_20 = fs_mult * 20;
+ int fs_mult_120 = fs_mult * 120;
+ int fs_mult_dist_len = fs_mult * kDistortionLength;
+ int fs_mult_lpc_analysis_len = fs_mult * kLpcAnalysisLength;
+
+ const size_t signal_length = 256 * fs_mult;
+ const int16_t* audio_history =
+ &(*sync_buffer_)[0][sync_buffer_->Size() - signal_length];
+
+ // Initialize some member variables.
+ lag_index_direction_ = 1;
+ current_lag_index_ = -1;
+ stop_muting_ = false;
+ random_vector_->set_seed_increment(1);
+ consecutive_expands_ = 0;
+ for (size_t ix = 0; ix < num_channels_; ++ix) {
+ channel_parameters_[ix].current_voice_mix_factor = 16384; // 1.0 in Q14.
+ channel_parameters_[ix].mute_factor = 16384; // 1.0 in Q14.
+ // Start with 0 gain for background noise.
+ background_noise_->SetMuteFactor(ix, 0);
+ }
+
+ // Calculate correlation in downsampled domain (4 kHz sample rate).
+ int16_t correlation_scale;
+ int correlation_length = Correlation(audio_history, signal_length,
+ correlation_vector, &correlation_scale);
+ correlation_length = 51; // TODO(hlundin): Legacy bit-exactness.
+
+ // Find peaks in correlation vector.
+ DspHelper::PeakDetection(correlation_vector, correlation_length,
+ kNumCorrelationCandidates, fs_mult,
+ best_correlation_index, best_correlation);
+
+ // Adjust peak locations; cross-correlation lags start at 2.5 ms
+ // (20 * fs_mult samples).
+ best_correlation_index[0] += fs_mult_20;
+ best_correlation_index[1] += fs_mult_20;
+ best_correlation_index[2] += fs_mult_20;
+
+ // Calculate distortion around the |kNumCorrelationCandidates| best lags.
+ int distortion_scale = 0;
+ for (int i = 0; i < kNumCorrelationCandidates; i++) {
+ int16_t min_index = std::max(fs_mult_20,
+ best_correlation_index[i] - fs_mult_4);
+ int16_t max_index = std::min(fs_mult_120 - 1,
+ best_correlation_index[i] + fs_mult_4);
+ best_distortion_index[i] = DspHelper::MinDistortion(
+ &(audio_history[signal_length - fs_mult_dist_len]), min_index,
+ max_index, fs_mult_dist_len, &best_distortion_w32[i]);
+ distortion_scale = std::max(16 - WebRtcSpl_NormW32(best_distortion_w32[i]),
+ distortion_scale);
+ }
+ // Shift the distortion values to fit in 16 bits.
+ WebRtcSpl_VectorBitShiftW32ToW16(best_distortion, kNumCorrelationCandidates,
+ best_distortion_w32, distortion_scale);
+
+ // Find the maximizing index |i| of the cost function
+ // f[i] = best_correlation[i] / best_distortion[i].
+ int32_t best_ratio = -1;
+ int best_index = -1;
+ for (int i = 0; i < kNumCorrelationCandidates; ++i) {
+ int32_t ratio;
+ if (best_distortion[i] > 0) {
+ ratio = (best_correlation[i] << 16) / best_distortion[i];
+ } else {
+ assert(best_correlation[i] == 0); // If one is zero, both must be.
+ ratio = 0; // Divide zero by zero => set result to zero.
+ }
+ if (ratio > best_ratio) {
+ best_index = i;
+ best_ratio = ratio;
+ }
+ }
+
+ int distortion_lag = best_distortion_index[best_index];
+ int correlation_lag = best_correlation_index[best_index];
+ max_lag_ = std::max(distortion_lag, correlation_lag);
+
+ // Calculate the exact best correlation in the range between
+ // |correlation_lag| and |distortion_lag|.
+ correlation_length = distortion_lag + 10;
+ correlation_length = std::min(correlation_length, fs_mult_120);
+ correlation_length = std::max(correlation_length, 60 * fs_mult);
+
+ int start_index = std::min(distortion_lag, correlation_lag);
+ int correlation_lags = WEBRTC_SPL_ABS_W16((distortion_lag-correlation_lag))
+ + 1;
+ assert(correlation_lags <= 99 * fs_mult + 1); // Cannot be larger.
+
+ for (size_t channel_ix = 0; channel_ix < num_channels_; ++channel_ix) {
+ ChannelParameters& parameters = channel_parameters_[channel_ix];
+ // Calculate suitable scaling.
+ int16_t signal_max = WebRtcSpl_MaxAbsValueW16(
+ &audio_history[signal_length - correlation_length - start_index
+ - correlation_lags],
+ correlation_length + start_index + correlation_lags - 1);
+ correlation_scale = ((31 - WebRtcSpl_NormW32(signal_max * signal_max))
+ + (31 - WebRtcSpl_NormW32(correlation_length))) - 31;
+ correlation_scale = std::max(static_cast<int16_t>(0), correlation_scale);
+
+ // Calculate the correlation, store in |correlation_vector2|.
+ WebRtcSpl_CrossCorrelation(
+ correlation_vector2,
+ &(audio_history[signal_length - correlation_length]),
+ &(audio_history[signal_length - correlation_length - start_index]),
+ correlation_length, correlation_lags, correlation_scale, -1);
+
+ // Find maximizing index.
+ best_index = WebRtcSpl_MaxIndexW32(correlation_vector2, correlation_lags);
+ int32_t max_correlation = correlation_vector2[best_index];
+ // Compensate index with start offset.
+ best_index = best_index + start_index;
+
+ // Calculate energies.
+ int32_t energy1 = WebRtcSpl_DotProductWithScale(
+ &(audio_history[signal_length - correlation_length]),
+ &(audio_history[signal_length - correlation_length]),
+ correlation_length, correlation_scale);
+ int32_t energy2 = WebRtcSpl_DotProductWithScale(
+ &(audio_history[signal_length - correlation_length - best_index]),
+ &(audio_history[signal_length - correlation_length - best_index]),
+ correlation_length, correlation_scale);
+
+ // Calculate the correlation coefficient between the two portions of the
+ // signal.
+ int16_t corr_coefficient;
+ if ((energy1 > 0) && (energy2 > 0)) {
+ int energy1_scale = std::max(16 - WebRtcSpl_NormW32(energy1), 0);
+ int energy2_scale = std::max(16 - WebRtcSpl_NormW32(energy2), 0);
+ // Make sure total scaling is even (to simplify scale factor after sqrt).
+ if ((energy1_scale + energy2_scale) & 1) {
+ // If sum is odd, add 1 to make it even.
+ energy1_scale += 1;
+ }
+ int16_t scaled_energy1 = energy1 >> energy1_scale;
+ int16_t scaled_energy2 = energy2 >> energy2_scale;
+ int16_t sqrt_energy_product = WebRtcSpl_SqrtFloor(
+ scaled_energy1 * scaled_energy2);
+ // Calculate max_correlation / sqrt(energy1 * energy2) in Q14.
+ int cc_shift = 14 - (energy1_scale + energy2_scale) / 2;
+ max_correlation = WEBRTC_SPL_SHIFT_W32(max_correlation, cc_shift);
+ corr_coefficient = WebRtcSpl_DivW32W16(max_correlation,
+ sqrt_energy_product);
+ corr_coefficient = std::min(static_cast<int16_t>(16384),
+ corr_coefficient); // Cap at 1.0 in Q14.
+ } else {
+ corr_coefficient = 0;
+ }
+
+ // Extract the two vectors expand_vector0 and expand_vector1 from
+ // |audio_history|.
+ int16_t expansion_length = max_lag_ + overlap_length_;
+ const int16_t* vector1 = &(audio_history[signal_length - expansion_length]);
+ const int16_t* vector2 = vector1 - distortion_lag;
+ // Normalize the second vector to the same energy as the first.
+ energy1 = WebRtcSpl_DotProductWithScale(vector1, vector1, expansion_length,
+ correlation_scale);
+ energy2 = WebRtcSpl_DotProductWithScale(vector2, vector2, expansion_length,
+ correlation_scale);
+ // Confirm that amplitude ratio sqrt(energy1 / energy2) is within 0.5 - 2.0,
+ // i.e., energy1 / energy1 is within 0.25 - 4.
+ int16_t amplitude_ratio;
+ if ((energy1 / 4 < energy2) && (energy1 > energy2 / 4)) {
+ // Energy constraint fulfilled. Use both vectors and scale them
+ // accordingly.
+ int16_t scaled_energy2 = std::max(16 - WebRtcSpl_NormW32(energy2), 0);
+ int16_t scaled_energy1 = scaled_energy2 - 13;
+ // Calculate scaled_energy1 / scaled_energy2 in Q13.
+ int32_t energy_ratio = WebRtcSpl_DivW32W16(
+ WEBRTC_SPL_SHIFT_W32(energy1, -scaled_energy1),
+ WEBRTC_SPL_RSHIFT_W32(energy2, scaled_energy2));
+ // Calculate sqrt ratio in Q13 (sqrt of en1/en2 in Q26).
+ amplitude_ratio = WebRtcSpl_SqrtFloor(energy_ratio << 13);
+ // Copy the two vectors and give them the same energy.
+ parameters.expand_vector0.Clear();
+ parameters.expand_vector0.PushBack(vector1, expansion_length);
+ parameters.expand_vector1.Clear();
+ if (parameters.expand_vector1.Size() <
+ static_cast<size_t>(expansion_length)) {
+ parameters.expand_vector1.Extend(
+ expansion_length - parameters.expand_vector1.Size());
+ }
+ WebRtcSpl_AffineTransformVector(¶meters.expand_vector1[0],
+ const_cast<int16_t*>(vector2),
+ amplitude_ratio,
+ 4096,
+ 13,
+ expansion_length);
+ } else {
+ // Energy change constraint not fulfilled. Only use last vector.
+ parameters.expand_vector0.Clear();
+ parameters.expand_vector0.PushBack(vector1, expansion_length);
+ // Copy from expand_vector0 to expand_vector1.
+ parameters.expand_vector0.CopyFrom(¶meters.expand_vector1);
+ // Set the energy_ratio since it is used by muting slope.
+ if ((energy1 / 4 < energy2) || (energy2 == 0)) {
+ amplitude_ratio = 4096; // 0.5 in Q13.
+ } else {
+ amplitude_ratio = 16384; // 2.0 in Q13.
+ }
+ }
+
+ // Set the 3 lag values.
+ int lag_difference = distortion_lag - correlation_lag;
+ if (lag_difference == 0) {
+ // |distortion_lag| and |correlation_lag| are equal.
+ expand_lags_[0] = distortion_lag;
+ expand_lags_[1] = distortion_lag;
+ expand_lags_[2] = distortion_lag;
+ } else {
+ // |distortion_lag| and |correlation_lag| are not equal; use different
+ // combinations of the two.
+ // First lag is |distortion_lag| only.
+ expand_lags_[0] = distortion_lag;
+ // Second lag is the average of the two.
+ expand_lags_[1] = (distortion_lag + correlation_lag) / 2;
+ // Third lag is the average again, but rounding towards |correlation_lag|.
+ if (lag_difference > 0) {
+ expand_lags_[2] = (distortion_lag + correlation_lag - 1) / 2;
+ } else {
+ expand_lags_[2] = (distortion_lag + correlation_lag + 1) / 2;
+ }
+ }
+
+ // Calculate the LPC and the gain of the filters.
+ // Calculate scale value needed for auto-correlation.
+ correlation_scale = WebRtcSpl_MaxAbsValueW16(
+ &(audio_history[signal_length - fs_mult_lpc_analysis_len]),
+ fs_mult_lpc_analysis_len);
+
+ correlation_scale = std::min(16 - WebRtcSpl_NormW32(correlation_scale), 0);
+ correlation_scale = std::max(correlation_scale * 2 + 7, 0);
+
+ // Calculate kUnvoicedLpcOrder + 1 lags of the auto-correlation function.
+ size_t temp_index = signal_length - fs_mult_lpc_analysis_len -
+ kUnvoicedLpcOrder;
+ // Copy signal to temporary vector to be able to pad with leading zeros.
+ int16_t* temp_signal = new int16_t[fs_mult_lpc_analysis_len
+ + kUnvoicedLpcOrder];
+ memset(temp_signal, 0,
+ sizeof(int16_t) * (fs_mult_lpc_analysis_len + kUnvoicedLpcOrder));
+ memcpy(&temp_signal[kUnvoicedLpcOrder],
+ &audio_history[temp_index + kUnvoicedLpcOrder],
+ sizeof(int16_t) * fs_mult_lpc_analysis_len);
+ WebRtcSpl_CrossCorrelation(auto_correlation,
+ &temp_signal[kUnvoicedLpcOrder],
+ &temp_signal[kUnvoicedLpcOrder],
+ fs_mult_lpc_analysis_len, kUnvoicedLpcOrder + 1,
+ correlation_scale, -1);
+ delete [] temp_signal;
+
+ // Verify that variance is positive.
+ if (auto_correlation[0] > 0) {
+ // Estimate AR filter parameters using Levinson-Durbin algorithm;
+ // kUnvoicedLpcOrder + 1 filter coefficients.
+ int16_t stability = WebRtcSpl_LevinsonDurbin(auto_correlation,
+ parameters.ar_filter,
+ reflection_coeff,
+ kUnvoicedLpcOrder);
+
+ // Keep filter parameters only if filter is stable.
+ if (stability != 1) {
+ // Set first coefficient to 4096 (1.0 in Q12).
+ parameters.ar_filter[0] = 4096;
+ // Set remaining |kUnvoicedLpcOrder| coefficients to zero.
+ WebRtcSpl_MemSetW16(parameters.ar_filter + 1, 0, kUnvoicedLpcOrder);
+ }
+ }
+
+ if (channel_ix == 0) {
+ // Extract a noise segment.
+ int16_t noise_length;
+ if (distortion_lag < 40) {
+ noise_length = 2 * distortion_lag + 30;
+ } else {
+ noise_length = distortion_lag + 30;
+ }
+ if (noise_length <= RandomVector::kRandomTableSize) {
+ memcpy(random_vector, RandomVector::kRandomTable,
+ sizeof(int16_t) * noise_length);
+ } else {
+ // Only applies to SWB where length could be larger than
+ // |kRandomTableSize|.
+ memcpy(random_vector, RandomVector::kRandomTable,
+ sizeof(int16_t) * RandomVector::kRandomTableSize);
+ assert(noise_length <= kMaxSampleRate / 8000 * 120 + 30);
+ random_vector_->IncreaseSeedIncrement(2);
+ random_vector_->Generate(
+ noise_length - RandomVector::kRandomTableSize,
+ &random_vector[RandomVector::kRandomTableSize]);
+ }
+ }
+
+ // Set up state vector and calculate scale factor for unvoiced filtering.
+ memcpy(parameters.ar_filter_state,
+ &(audio_history[signal_length - kUnvoicedLpcOrder]),
+ sizeof(int16_t) * kUnvoicedLpcOrder);
+ memcpy(unvoiced_vector - kUnvoicedLpcOrder,
+ &(audio_history[signal_length - 128 - kUnvoicedLpcOrder]),
+ sizeof(int16_t) * kUnvoicedLpcOrder);
+ WebRtcSpl_FilterMAFastQ12(
+ const_cast<int16_t*>(&audio_history[signal_length - 128]),
+ unvoiced_vector, parameters.ar_filter, kUnvoicedLpcOrder + 1, 128);
+ int16_t unvoiced_prescale;
+ if (WebRtcSpl_MaxAbsValueW16(unvoiced_vector, 128) > 4000) {
+ unvoiced_prescale = 4;
+ } else {
+ unvoiced_prescale = 0;
+ }
+ int32_t unvoiced_energy = WebRtcSpl_DotProductWithScale(unvoiced_vector,
+ unvoiced_vector,
+ 128,
+ unvoiced_prescale);
+
+ // Normalize |unvoiced_energy| to 28 or 29 bits to preserve sqrt() accuracy.
+ int16_t unvoiced_scale = WebRtcSpl_NormW32(unvoiced_energy) - 3;
+ // Make sure we do an odd number of shifts since we already have 7 shifts
+ // from dividing with 128 earlier. This will make the total scale factor
+ // even, which is suitable for the sqrt.
+ unvoiced_scale += ((unvoiced_scale & 0x1) ^ 0x1);
+ unvoiced_energy = WEBRTC_SPL_SHIFT_W32(unvoiced_energy, unvoiced_scale);
+ int32_t unvoiced_gain = WebRtcSpl_SqrtFloor(unvoiced_energy);
+ parameters.ar_gain_scale = 13
+ + (unvoiced_scale + 7 - unvoiced_prescale) / 2;
+ parameters.ar_gain = unvoiced_gain;
+
+ // Calculate voice_mix_factor from corr_coefficient.
+ // Let x = corr_coefficient. Then, we compute:
+ // if (x > 0.48)
+ // voice_mix_factor = (-5179 + 19931x - 16422x^2 + 5776x^3) / 4096;
+ // else
+ // voice_mix_factor = 0;
+ if (corr_coefficient > 7875) {
+ int16_t x1, x2, x3;
+ x1 = corr_coefficient; // |corr_coefficient| is in Q14.
+ x2 = (x1 * x1) >> 14; // Shift 14 to keep result in Q14.
+ x3 = (x1 * x2) >> 14;
+ static const int kCoefficients[4] = { -5179, 19931, -16422, 5776 };
+ int32_t temp_sum = kCoefficients[0] << 14;
+ temp_sum += kCoefficients[1] * x1;
+ temp_sum += kCoefficients[2] * x2;
+ temp_sum += kCoefficients[3] * x3;
+ parameters.voice_mix_factor = temp_sum / 4096;
+ parameters.voice_mix_factor = std::min(parameters.voice_mix_factor,
+ static_cast<int16_t>(16384));
+ parameters.voice_mix_factor = std::max(parameters.voice_mix_factor,
+ static_cast<int16_t>(0));
+ } else {
+ parameters.voice_mix_factor = 0;
+ }
+
+ // Calculate muting slope. Reuse value from earlier scaling of
+ // |expand_vector0| and |expand_vector1|.
+ int16_t slope = amplitude_ratio;
+ if (slope > 12288) {
+ // slope > 1.5.
+ // Calculate (1 - (1 / slope)) / distortion_lag =
+ // (slope - 1) / (distortion_lag * slope).
+ // |slope| is in Q13, so 1 corresponds to 8192. Shift up to Q25 before
+ // the division.
+ // Shift the denominator from Q13 to Q5 before the division. The result of
+ // the division will then be in Q20.
+ int16_t temp_ratio = WebRtcSpl_DivW32W16((slope - 8192) << 12,
+ (distortion_lag * slope) >> 8);
+ if (slope > 14746) {
+ // slope > 1.8.
+ // Divide by 2, with proper rounding.
+ parameters.mute_slope = (temp_ratio + 1) / 2;
+ } else {
+ // Divide by 8, with proper rounding.
+ parameters.mute_slope = (temp_ratio + 4) / 8;
+ }
+ parameters.onset = true;
+ } else {
+ // Calculate (1 - slope) / distortion_lag.
+ // Shift |slope| by 7 to Q20 before the division. The result is in Q20.
+ parameters.mute_slope = WebRtcSpl_DivW32W16((8192 - slope) << 7,
+ distortion_lag);
+ if (parameters.voice_mix_factor <= 13107) {
+ // Make sure the mute factor decreases from 1.0 to 0.9 in no more than
+ // 6.25 ms.
+ // mute_slope >= 0.005 / fs_mult in Q20.
+ parameters.mute_slope = std::max(static_cast<int16_t>(5243 / fs_mult),
+ parameters.mute_slope);
+ } else if (slope > 8028) {
+ parameters.mute_slope = 0;
+ }
+ parameters.onset = false;
+ }
+ }
+}
+
+int16_t Expand::Correlation(const int16_t* input, int16_t input_length,
+ int16_t* output, int16_t* output_scale) const {
+ // Set parameters depending on sample rate.
+ const int16_t* filter_coefficients;
+ int16_t num_coefficients;
+ int16_t downsampling_factor;
+ if (fs_hz_ == 8000) {
+ num_coefficients = 3;
+ downsampling_factor = 2;
+ filter_coefficients = DspHelper::kDownsample8kHzTbl;
+ } else if (fs_hz_ == 16000) {
+ num_coefficients = 5;
+ downsampling_factor = 4;
+ filter_coefficients = DspHelper::kDownsample16kHzTbl;
+ } else if (fs_hz_ == 32000) {
+ num_coefficients = 7;
+ downsampling_factor = 8;
+ filter_coefficients = DspHelper::kDownsample32kHzTbl;
+ } else { // fs_hz_ == 48000.
+ num_coefficients = 7;
+ downsampling_factor = 12;
+ filter_coefficients = DspHelper::kDownsample48kHzTbl;
+ }
+
+ // Correlate from lag 10 to lag 60 in downsampled domain.
+ // (Corresponds to 20-120 for narrow-band, 40-240 for wide-band, and so on.)
+ static const int kCorrelationStartLag = 10;
+ static const int kNumCorrelationLags = 54;
+ static const int kCorrelationLength = 60;
+ // Downsample to 4 kHz sample rate.
+ static const int kDownsampledLength = kCorrelationStartLag
+ + kNumCorrelationLags + kCorrelationLength;
+ int16_t downsampled_input[kDownsampledLength];
+ static const int kFilterDelay = 0;
+ WebRtcSpl_DownsampleFast(
+ input + input_length - kDownsampledLength * downsampling_factor,
+ kDownsampledLength * downsampling_factor, downsampled_input,
+ kDownsampledLength, filter_coefficients, num_coefficients,
+ downsampling_factor, kFilterDelay);
+
+ // Normalize |downsampled_input| to using all 16 bits.
+ int16_t max_value = WebRtcSpl_MaxAbsValueW16(downsampled_input,
+ kDownsampledLength);
+ int16_t norm_shift = 16 - WebRtcSpl_NormW32(max_value);
+ WebRtcSpl_VectorBitShiftW16(downsampled_input, kDownsampledLength,
+ downsampled_input, norm_shift);
+
+ int32_t correlation[kNumCorrelationLags];
+ static const int kCorrelationShift = 6;
+ WebRtcSpl_CrossCorrelation(
+ correlation,
+ &downsampled_input[kDownsampledLength - kCorrelationLength],
+ &downsampled_input[kDownsampledLength - kCorrelationLength
+ - kCorrelationStartLag],
+ kCorrelationLength, kNumCorrelationLags, kCorrelationShift, -1);
+
+ // Normalize and move data from 32-bit to 16-bit vector.
+ int32_t max_correlation = WebRtcSpl_MaxAbsValueW32(correlation,
+ kNumCorrelationLags);
+ int16_t norm_shift2 = std::max(18 - WebRtcSpl_NormW32(max_correlation), 0);
+ WebRtcSpl_VectorBitShiftW32ToW16(output, kNumCorrelationLags, correlation,
+ norm_shift2);
+ // Total scale factor (right shifts) of correlation value.
+ *output_scale = 2 * norm_shift + kCorrelationShift + norm_shift2;
+ return kNumCorrelationLags;
+}
+
+void Expand::UpdateLagIndex() {
+ current_lag_index_ = current_lag_index_ + lag_index_direction_;
+ // Change direction if needed.
+ if (current_lag_index_ <= 0) {
+ lag_index_direction_ = 1;
+ }
+ if (current_lag_index_ >= kNumLags - 1) {
+ lag_index_direction_ = -1;
+ }
+}
+
+} // namespace webrtc