mirror of
https://github.com/oxen-io/session-android.git
synced 2024-11-28 20:45:17 +00:00
217 lines
8.5 KiB
C++
217 lines
8.5 KiB
C++
/*
|
|
* 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
|