51c7cbb86a
This reverts commit 518c683f3e413523a458a94b533274bd7f29992d. Breaks Linux-Asan bot https://uberchromegw.corp.google.com/i/client.webrtc/builders/Linux%20Asan/builds/4348/steps/libjingle_peerconnection_unittest/logs/stdio BUG= TBR=aluebs@webrtc.org Review URL: https://codereview.webrtc.org/1208793002. Cr-Commit-Position: refs/heads/master@{#9503}
84 lines
2.8 KiB
C++
84 lines
2.8 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_AGC_AGC_AUDIO_PROC_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_AUDIO_PROC_H_
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/modules/audio_processing/agc/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 AgcAudioProc {
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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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AgcAudioProc();
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~AgcAudioProc();
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int ExtractFeatures(const int16_t* audio_frame,
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int length,
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AudioFeatures* audio_features);
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static const int kDftSize = 512;
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private:
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void PitchAnalysis(double* pitch_gains, double* pitch_lags_hz, int length);
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void SubframeCorrelation(double* corr, int length_corr, int subframe_index);
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void GetLpcPolynomials(double* lpc, int length_lpc);
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void FindFirstSpectralPeaks(double* f_peak, int length_f_peak);
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void Rms(double* rms, int 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 int kNumPastSignalSamples = 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 int kNum10msSubframes = 3;
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static const int kNumSubframeSamples = kSampleRateHz / 100;
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static const int kNumSamplesToProcess = kNum10msSubframes *
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kNumSubframeSamples; // Samples in 30 ms @ given sampling rate.
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static const int kBufferLength = kNumPastSignalSamples + kNumSamplesToProcess;
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static const int kIpLength = kDftSize >> 1;
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static const int kWLength = kDftSize >> 1;
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static const int kLpcOrder = 16;
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int 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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int 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_AGC_AGC_AUDIO_PROC_H_
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