753eada3aa
Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1 Update notes: * Pull in third party license file * Replace .gypi files with BUILD.gn to keep track of what changes upstream * Bunch of new filse pulled in as dependencies * Won't build yet due to changes needed on top of these
90 lines
3.0 KiB
C++
90 lines
3.0 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/modules/audio_processing/vad/common.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class AudioFrame;
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class PoleZeroFilter;
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class VadAudioProc {
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public:
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// Forward declare iSAC structs.
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struct PitchAnalysisStruct;
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struct PreFiltBankstr;
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VadAudioProc();
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~VadAudioProc();
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int ExtractFeatures(const int16_t* audio_frame,
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size_t length,
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AudioFeatures* audio_features);
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static const size_t kDftSize = 512;
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private:
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void PitchAnalysis(double* pitch_gains, double* pitch_lags_hz, size_t length);
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void SubframeCorrelation(double* corr,
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size_t length_corr,
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size_t subframe_index);
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void GetLpcPolynomials(double* lpc, size_t length_lpc);
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void FindFirstSpectralPeaks(double* f_peak, size_t length_f_peak);
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void Rms(double* rms, size_t length_rms);
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void ResetBuffer();
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// To compute spectral peak we perform LPC analysis to get spectral envelope.
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// For every 30 ms we compute 3 spectral peak there for 3 LPC analysis.
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// LPC is computed over 15 ms of windowed audio. For every 10 ms sub-frame
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// we need 5 ms of past signal to create the input of LPC analysis.
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static const size_t kNumPastSignalSamples =
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static_cast<size_t>(kSampleRateHz / 200);
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// TODO(turajs): maybe defining this at a higher level (maybe enum) so that
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// all the code recognize it as "no-error."
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static const int kNoError = 0;
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static const size_t kNum10msSubframes = 3;
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static const size_t kNumSubframeSamples =
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static_cast<size_t>(kSampleRateHz / 100);
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static const size_t kNumSamplesToProcess =
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kNum10msSubframes *
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kNumSubframeSamples; // Samples in 30 ms @ given sampling rate.
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static const size_t kBufferLength =
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kNumPastSignalSamples + kNumSamplesToProcess;
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static const size_t kIpLength = kDftSize >> 1;
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static const size_t kWLength = kDftSize >> 1;
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static const size_t kLpcOrder = 16;
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size_t ip_[kIpLength];
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float w_fft_[kWLength];
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// A buffer of 5 ms (past audio) + 30 ms (one iSAC frame ).
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float audio_buffer_[kBufferLength];
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size_t num_buffer_samples_;
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double log_old_gain_;
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double old_lag_;
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rtc::scoped_ptr<PitchAnalysisStruct> pitch_analysis_handle_;
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rtc::scoped_ptr<PreFiltBankstr> pre_filter_handle_;
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rtc::scoped_ptr<PoleZeroFilter> high_pass_filter_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_
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