| OLD | NEW |
|---|
| (Empty) |
| 1 /* | |
| 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | |
| 3 * | |
| 4 * Use of this source code is governed by a BSD-style license | |
| 5 * that can be found in the LICENSE file in the root of the source | |
| 6 * tree. An additional intellectual property rights grant can be found | |
| 7 * in the file PATENTS. All contributing project authors may | |
| 8 * be found in the AUTHORS file in the root of the source tree. | |
| 9 */ | |
| 10 | |
| 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_ | |
| 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_ | |
| 13 | |
| 14 #include "webrtc/base/scoped_ptr.h" | |
| 15 #include "webrtc/modules/audio_processing/vad/common.h" | |
| 16 #include "webrtc/typedefs.h" | |
| 17 | |
| 18 namespace webrtc { | |
| 19 | |
| 20 class AudioFrame; | |
| 21 class PoleZeroFilter; | |
| 22 | |
| 23 class VadAudioProc { | |
| 24 public: | |
| 25 // Forward declare iSAC structs. | |
| 26 struct PitchAnalysisStruct; | |
| 27 struct PreFiltBankstr; | |
| 28 | |
| 29 VadAudioProc(); | |
| 30 ~VadAudioProc(); | |
| 31 | |
| 32 int ExtractFeatures(const int16_t* audio_frame, | |
| 33 int length, | |
| 34 AudioFeatures* audio_features); | |
| 35 | |
| 36 static const int kDftSize = 512; | |
| 37 | |
| 38 private: | |
| 39 void PitchAnalysis(double* pitch_gains, double* pitch_lags_hz, int length); | |
| 40 void SubframeCorrelation(double* corr, int length_corr, int subframe_index); | |
| 41 void GetLpcPolynomials(double* lpc, int length_lpc); | |
| 42 void FindFirstSpectralPeaks(double* f_peak, int length_f_peak); | |
| 43 void Rms(double* rms, int length_rms); | |
| 44 void ResetBuffer(); | |
| 45 | |
| 46 // To compute spectral peak we perform LPC analysis to get spectral envelope. | |
| 47 // For every 30 ms we compute 3 spectral peak there for 3 LPC analysis. | |
| 48 // LPC is computed over 15 ms of windowed audio. For every 10 ms sub-frame | |
| 49 // we need 5 ms of past signal to create the input of LPC analysis. | |
| 50 static const int kNumPastSignalSamples = kSampleRateHz / 200; | |
| 51 | |
| 52 // TODO(turajs): maybe defining this at a higher level (maybe enum) so that | |
| 53 // all the code recognize it as "no-error." | |
| 54 static const int kNoError = 0; | |
| 55 | |
| 56 static const int kNum10msSubframes = 3; | |
| 57 static const int kNumSubframeSamples = kSampleRateHz / 100; | |
| 58 static const int kNumSamplesToProcess = | |
| 59 kNum10msSubframes * | |
| 60 kNumSubframeSamples; // Samples in 30 ms @ given sampling rate. | |
| 61 static const int kBufferLength = kNumPastSignalSamples + kNumSamplesToProcess; | |
| 62 static const int kIpLength = kDftSize >> 1; | |
| 63 static const int kWLength = kDftSize >> 1; | |
| 64 | |
| 65 static const int kLpcOrder = 16; | |
| 66 | |
| 67 int ip_[kIpLength]; | |
| 68 float w_fft_[kWLength]; | |
| 69 | |
| 70 // A buffer of 5 ms (past audio) + 30 ms (one iSAC frame ). | |
| 71 float audio_buffer_[kBufferLength]; | |
| 72 int num_buffer_samples_; | |
| 73 | |
| 74 double log_old_gain_; | |
| 75 double old_lag_; | |
| 76 | |
| 77 rtc::scoped_ptr<PitchAnalysisStruct> pitch_analysis_handle_; | |
| 78 rtc::scoped_ptr<PreFiltBankstr> pre_filter_handle_; | |
| 79 rtc::scoped_ptr<PoleZeroFilter> high_pass_filter_; | |
| 80 }; | |
| 81 | |
| 82 } // namespace webrtc | |
| 83 | |
| 84 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_H_ | |
| OLD | NEW |
|---|
Chromium Code Reviews
Issue 1208793002:
Revert "Pull the Voice Activity Detector out from the AGC" (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master