webrtc/modules/audio_processing/vad/vad_audio_proc.h
Peter Kasting 662d7f11d5 Fixes to support building in -std=c++20 mode.
* Structs with user-declared constructors are no longer considered
  aggregates, so remove the declarations when possible
* Types of both arguments to "==" must match to avoid "ambiguous
  function call" warning
* Various types of math involving enums are deprecated, so replace with
  constexprs where necessary
* ABSL_CONST_INIT must be used on definition as well as declaration
* volatile memory may no longer be read from and written to by the same
  operator, so replace e.g. "n++" with "n = n + 1"
* Replace an outdated check for no_unique_address support with
  __has_cpp_attribute
* std::result_of(f(x)) has been removed, replace with
  std::invoke_result(f, x)

Bug: chromium:1284275
Change-Id: I77b366ab1da7eb2c1e4c825b2714417c31ee5903
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261221
Auto-Submit: Peter Kasting <pkasting@chromium.org>
Reviewed-by: Tomas Gunnarsson <tommi@google.com>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36786}
2022-05-05 17:15:58 +00:00

90 lines
3 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.
*/
#ifndef MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_
#define MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_
#include <stddef.h>
#include <stdint.h>
#include <memory>
#include "modules/audio_processing/vad/common.h" // AudioFeatures, kSampleR...
namespace webrtc {
class PoleZeroFilter;
class VadAudioProc {
public:
// Forward declare iSAC structs.
struct PitchAnalysisStruct;
struct PreFiltBankstr;
VadAudioProc();
~VadAudioProc();
int ExtractFeatures(const int16_t* audio_frame,
size_t length,
AudioFeatures* audio_features);
static constexpr size_t kDftSize = 512;
private:
void PitchAnalysis(double* pitch_gains, double* pitch_lags_hz, size_t length);
void SubframeCorrelation(double* corr,
size_t length_corr,
size_t subframe_index);
void GetLpcPolynomials(double* lpc, size_t length_lpc);
void FindFirstSpectralPeaks(double* f_peak, size_t length_f_peak);
void Rms(double* rms, size_t length_rms);
void ResetBuffer();
// To compute spectral peak we perform LPC analysis to get spectral envelope.
// For every 30 ms we compute 3 spectral peak there for 3 LPC analysis.
// LPC is computed over 15 ms of windowed audio. For every 10 ms sub-frame
// we need 5 ms of past signal to create the input of LPC analysis.
static constexpr size_t kNumPastSignalSamples =
static_cast<size_t>(kSampleRateHz / 200);
// TODO(turajs): maybe defining this at a higher level (maybe enum) so that
// all the code recognize it as "no-error."
static constexpr int kNoError = 0;
static constexpr size_t kNum10msSubframes = 3;
static constexpr size_t kNumSubframeSamples =
static_cast<size_t>(kSampleRateHz / 100);
// Samples in 30 ms @ given sampling rate.
static constexpr size_t kNumSamplesToProcess =
size_t{kNum10msSubframes} * kNumSubframeSamples;
static constexpr size_t kBufferLength =
size_t{kNumPastSignalSamples} + kNumSamplesToProcess;
static constexpr size_t kIpLength = kDftSize >> 1;
static constexpr size_t kWLength = kDftSize >> 1;
static constexpr size_t kLpcOrder = 16;
size_t ip_[kIpLength];
float w_fft_[kWLength];
// A buffer of 5 ms (past audio) + 30 ms (one iSAC frame ).
float audio_buffer_[kBufferLength];
size_t num_buffer_samples_;
double log_old_gain_;
double old_lag_;
std::unique_ptr<PitchAnalysisStruct> pitch_analysis_handle_;
std::unique_ptr<PreFiltBankstr> pre_filter_handle_;
std::unique_ptr<PoleZeroFilter> high_pass_filter_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_