Rename neteq4 folder to neteq
Keep the old neteq4/audio_decoder_unittests.isolate while waiting for
a hard-coded reference to change.
This CL effectively reverts r6257 "Rename neteq4 folder to neteq".
BUG=2996
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6367 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq/time_stretch.cc b/webrtc/modules/audio_coding/neteq/time_stretch.cc
new file mode 100644
index 0000000..a9228d4
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq/time_stretch.cc
@@ -0,0 +1,216 @@
+/*
+ * 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 <algorithm> // min, max
+
+#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"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+
+namespace webrtc {
+
+TimeStretch::ReturnCodes TimeStretch::Process(
+ const int16_t* input,
+ size_t input_len,
+ AudioMultiVector* output,
+ int16_t* length_change_samples) {
+
+ // Pre-calculate common multiplication with |fs_mult_|.
+ int fs_mult_120 = fs_mult_ * 120; // Corresponds to 15 ms.
+
+ const int16_t* signal;
+ scoped_ptr<int16_t[]> 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,
+ static_cast<int>(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 int kNumPeaks = 1;
+ int 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 >= 0);
+ 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 >= 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(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<int16_t>(vec1_energy >> energy1_scale);
+ int16_t vec2_energy_int16 =
+ static_cast<int16_t>(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<int16_t>(16384), best_correlation);
+ }
+
+
+ // Check accelerate criteria and stretch the signal.
+ ReturnCodes return_value = CheckCriteriaAndStretch(
+ input, input_len, peak_index, best_correlation, active_speech, 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,
+ int 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 = 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