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/*
* 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 "modules/audio_processing/vad/vad_audio_proc.h"
#include <math.h>
#include <stdio.h>
#include <string.h>
#include "common_audio/third_party/fft4g/fft4g.h"
#include "modules/audio_processing/vad/pitch_internal.h"
#include "modules/audio_processing/vad/pole_zero_filter.h"
#include "modules/audio_processing/vad/vad_audio_proc_internal.h"
#include "rtc_base/checks.h"
extern "C" {
#include "modules/audio_coding/codecs/isac/main/source/filter_functions.h"
#include "modules/audio_coding/codecs/isac/main/source/isac_vad.h"
#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
#include "modules/audio_coding/codecs/isac/main/source/structs.h"
}
namespace webrtc {
// The following structures are declared anonymous in iSAC's structs.h. To
// forward declare them, we use this derived class trick.
struct VadAudioProc::PitchAnalysisStruct : public ::PitchAnalysisStruct {};
struct VadAudioProc::PreFiltBankstr : public ::PreFiltBankstr {};
static constexpr float kFrequencyResolution =
kSampleRateHz / static_cast<float>(VadAudioProc::kDftSize);
static constexpr int kSilenceRms = 5;
// TODO(turajs): Make a Create or Init for VadAudioProc.
VadAudioProc::VadAudioProc()
: audio_buffer_(),
num_buffer_samples_(kNumPastSignalSamples),
log_old_gain_(-2),
old_lag_(50), // Arbitrary but valid as pitch-lag (in samples).
pitch_analysis_handle_(new PitchAnalysisStruct),
pre_filter_handle_(new PreFiltBankstr),
high_pass_filter_(PoleZeroFilter::Create(kCoeffNumerator,
kFilterOrder,
kCoeffDenominator,
kFilterOrder)) {
static_assert(kNumPastSignalSamples + kNumSubframeSamples ==
sizeof(kLpcAnalWin) / sizeof(kLpcAnalWin[0]),
"lpc analysis window incorrect size");
static_assert(kLpcOrder + 1 == sizeof(kCorrWeight) / sizeof(kCorrWeight[0]),
"correlation weight incorrect size");
// TODO(turajs): Are we doing too much in the constructor?
float data[kDftSize];
// Make FFT to initialize.
ip_[0] = 0;
WebRtc_rdft(kDftSize, 1, data, ip_, w_fft_);
// TODO(turajs): Need to initialize high-pass filter.
// Initialize iSAC components.
WebRtcIsac_InitPreFilterbank(pre_filter_handle_.get());
WebRtcIsac_InitPitchAnalysis(pitch_analysis_handle_.get());
}
VadAudioProc::~VadAudioProc() {}
void VadAudioProc::ResetBuffer() {
memcpy(audio_buffer_, &audio_buffer_[kNumSamplesToProcess],
sizeof(audio_buffer_[0]) * kNumPastSignalSamples);
num_buffer_samples_ = kNumPastSignalSamples;
}
int VadAudioProc::ExtractFeatures(const int16_t* frame,
size_t length,
AudioFeatures* features) {
features->num_frames = 0;
if (length != kNumSubframeSamples) {
return -1;
}
// High-pass filter to remove the DC component and very low frequency content.
// We have experienced that this high-pass filtering improves voice/non-voiced
// classification.
if (high_pass_filter_->Filter(frame, kNumSubframeSamples,
&audio_buffer_[num_buffer_samples_]) != 0) {
return -1;
}
num_buffer_samples_ += kNumSubframeSamples;
if (num_buffer_samples_ < kBufferLength) {
return 0;
}
RTC_DCHECK_EQ(num_buffer_samples_, kBufferLength);
features->num_frames = kNum10msSubframes;
features->silence = false;
Rms(features->rms, kMaxNumFrames);
for (size_t i = 0; i < kNum10msSubframes; ++i) {
if (features->rms[i] < kSilenceRms) {
// PitchAnalysis can cause NaNs in the pitch gain if it's fed silence.
// Bail out here instead.
features->silence = true;
ResetBuffer();
return 0;
}
}
PitchAnalysis(features->log_pitch_gain, features->pitch_lag_hz,
kMaxNumFrames);
FindFirstSpectralPeaks(features->spectral_peak, kMaxNumFrames);
ResetBuffer();
return 0;
}
// Computes |kLpcOrder + 1| correlation coefficients.
void VadAudioProc::SubframeCorrelation(double* corr,
size_t length_corr,
size_t subframe_index) {
RTC_DCHECK_GE(length_corr, kLpcOrder + 1);
double windowed_audio[kNumSubframeSamples + kNumPastSignalSamples];
size_t buffer_index = subframe_index * kNumSubframeSamples;
for (size_t n = 0; n < kNumSubframeSamples + kNumPastSignalSamples; n++)
windowed_audio[n] = audio_buffer_[buffer_index++] * kLpcAnalWin[n];
WebRtcIsac_AutoCorr(corr, windowed_audio,
kNumSubframeSamples + kNumPastSignalSamples, kLpcOrder);
}
// Compute |kNum10msSubframes| sets of LPC coefficients, one per 10 ms input.
// The analysis window is 15 ms long and it is centered on the first half of
// each 10ms sub-frame. This is equivalent to computing LPC coefficients for the
// first half of each 10 ms subframe.
void VadAudioProc::GetLpcPolynomials(double* lpc, size_t length_lpc) {
RTC_DCHECK_GE(length_lpc, kNum10msSubframes * (kLpcOrder + 1));
double corr[kLpcOrder + 1];
double reflec_coeff[kLpcOrder];
for (size_t i = 0, offset_lpc = 0; i < kNum10msSubframes;
i++, offset_lpc += kLpcOrder + 1) {
SubframeCorrelation(corr, kLpcOrder + 1, i);
corr[0] *= 1.0001;
// This makes Lev-Durb a bit more stable.
for (size_t k = 0; k < kLpcOrder + 1; k++) {
corr[k] *= kCorrWeight[k];
}
WebRtcIsac_LevDurb(&lpc[offset_lpc], reflec_coeff, corr, kLpcOrder);
}
}
// Fit a second order curve to these 3 points and find the location of the
// extremum. The points are inverted before curve fitting.
static float QuadraticInterpolation(float prev_val,
float curr_val,
float next_val) {
// Doing the interpolation in |1 / A(z)|^2.
float fractional_index = 0;
next_val = 1.0f / next_val;
prev_val = 1.0f / prev_val;
curr_val = 1.0f / curr_val;
fractional_index =
-(next_val - prev_val) * 0.5f / (next_val + prev_val - 2.f * curr_val);
RTC_DCHECK_LT(fabs(fractional_index), 1);
return fractional_index;
}
// 1 / A(z), where A(z) is defined by |lpc| is a model of the spectral envelope
// of the input signal. The local maximum of the spectral envelope corresponds
// with the local minimum of A(z). It saves complexity, as we save one
// inversion. Furthermore, we find the first local maximum of magnitude squared,
// to save on one square root.
void VadAudioProc::FindFirstSpectralPeaks(double* f_peak,
size_t length_f_peak) {
RTC_DCHECK_GE(length_f_peak, kNum10msSubframes);
double lpc[kNum10msSubframes * (kLpcOrder + 1)];
// For all sub-frames.
GetLpcPolynomials(lpc, kNum10msSubframes * (kLpcOrder + 1));
const size_t kNumDftCoefficients = kDftSize / 2 + 1;
float data[kDftSize];
for (size_t i = 0; i < kNum10msSubframes; i++) {
// Convert to float with zero pad.
memset(data, 0, sizeof(data));
for (size_t n = 0; n < kLpcOrder + 1; n++) {
data[n] = static_cast<float>(lpc[i * (kLpcOrder + 1) + n]);
}
// Transform to frequency domain.
WebRtc_rdft(kDftSize, 1, data, ip_, w_fft_);
size_t index_peak = 0;
float prev_magn_sqr = data[0] * data[0];
float curr_magn_sqr = data[2] * data[2] + data[3] * data[3];
float next_magn_sqr;
bool found_peak = false;
for (size_t n = 2; n < kNumDftCoefficients - 1; n++) {
next_magn_sqr =
data[2 * n] * data[2 * n] + data[2 * n + 1] * data[2 * n + 1];
if (curr_magn_sqr < prev_magn_sqr && curr_magn_sqr < next_magn_sqr) {
found_peak = true;
index_peak = n - 1;
break;
}
prev_magn_sqr = curr_magn_sqr;
curr_magn_sqr = next_magn_sqr;
}
float fractional_index = 0;
if (!found_peak) {
// Checking if |kNumDftCoefficients - 1| is the local minimum.
next_magn_sqr = data[1] * data[1];
if (curr_magn_sqr < prev_magn_sqr && curr_magn_sqr < next_magn_sqr) {
index_peak = kNumDftCoefficients - 1;
}
} else {
// A peak is found, do a simple quadratic interpolation to get a more
// accurate estimate of the peak location.
fractional_index =
QuadraticInterpolation(prev_magn_sqr, curr_magn_sqr, next_magn_sqr);
}
f_peak[i] = (index_peak + fractional_index) * kFrequencyResolution;
}
}
// Using iSAC functions to estimate pitch gains & lags.
void VadAudioProc::PitchAnalysis(double* log_pitch_gains,
double* pitch_lags_hz,
size_t length) {
// TODO(turajs): This can be "imported" from iSAC & and the next two
// constants.
RTC_DCHECK_GE(length, kNum10msSubframes);
const int kNumPitchSubframes = 4;
double gains[kNumPitchSubframes];
double lags[kNumPitchSubframes];
const int kNumSubbandFrameSamples = 240;
const int kNumLookaheadSamples = 24;
float lower[kNumSubbandFrameSamples];
float upper[kNumSubbandFrameSamples];
double lower_lookahead[kNumSubbandFrameSamples];
double upper_lookahead[kNumSubbandFrameSamples];
double lower_lookahead_pre_filter[kNumSubbandFrameSamples +
kNumLookaheadSamples];
// Split signal to lower and upper bands
WebRtcIsac_SplitAndFilterFloat(&audio_buffer_[kNumPastSignalSamples], lower,
upper, lower_lookahead, upper_lookahead,
pre_filter_handle_.get());
WebRtcIsac_PitchAnalysis(lower_lookahead, lower_lookahead_pre_filter,
pitch_analysis_handle_.get(), lags, gains);
// Lags are computed on lower-band signal with sampling rate half of the
// input signal.
GetSubframesPitchParameters(
kSampleRateHz / 2, gains, lags, kNumPitchSubframes, kNum10msSubframes,
&log_old_gain_, &old_lag_, log_pitch_gains, pitch_lags_hz);
}
void VadAudioProc::Rms(double* rms, size_t length_rms) {
RTC_DCHECK_GE(length_rms, kNum10msSubframes);
size_t offset = kNumPastSignalSamples;
for (size_t i = 0; i < kNum10msSubframes; i++) {
rms[i] = 0;
for (size_t n = 0; n < kNumSubframeSamples; n++, offset++)
rms[i] += audio_buffer_[offset] * audio_buffer_[offset];
rms[i] = sqrt(rms[i] / kNumSubframeSamples);
}
}
} // namespace webrtc