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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.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_AUDIO_PROC_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_AUDIO_PROC_H_
#include "webrtc/modules/audio_processing/agc/common.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class AudioFrame;
class PoleZeroFilter;
class AgcAudioProc {
public:
// Forward declare iSAC structs.
struct PitchAnalysisStruct;
struct PreFiltBankstr;
AgcAudioProc();
~AgcAudioProc();
int ExtractFeatures(const int16_t* audio_frame,
int length,
AudioFeatures* audio_features);
static const int kDftSize = 512;
private:
void PitchAnalysis(double* pitch_gains, double* pitch_lags_hz, int length);
void SubframeCorrelation(double* corr, int lenght_corr, int subframe_index);
void GetLpcPolynomials(double* lpc, int length_lpc);
void FindFirstSpectralPeaks(double* f_peak, int length_f_peak);
void Rms(double* rms, int 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 const int kNumPastSignalSamples = kSampleRateHz / 200;
// TODO(turajs): maybe defining this at a higher level (maybe enum) so that
// all the code recognize it as "no-error."
static const int kNoError = 0;
static const int kNum10msSubframes = 3;
static const int kNumSubframeSamples = kSampleRateHz / 100;
static const int kNumSamplesToProcess = kNum10msSubframes *
kNumSubframeSamples; // Samples in 30 ms @ given sampling rate.
static const int kBufferLength = kNumPastSignalSamples + kNumSamplesToProcess;
static const int kIpLength = kDftSize >> 1;
static const int kWLength = kDftSize >> 1;
static const int kLpcOrder = 16;
int ip_[kIpLength];
float w_fft_[kWLength];
// A buffer of 5 ms (past audio) + 30 ms (one iSAC frame ).
float audio_buffer_[kBufferLength];
int num_buffer_samples_;
double log_old_gain_;
double old_lag_;
scoped_ptr<PitchAnalysisStruct> pitch_analysis_handle_;
scoped_ptr<PreFiltBankstr> pre_filter_handle_;
scoped_ptr<PoleZeroFilter> high_pass_filter_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_AUDIO_PROC_H_
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