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/*
* SPDX-FileCopyrightText: Copyright 2021-2022 Arm Limited and/or its affiliates <open-source-office@arm.com>
* SPDX-License-Identifier: Apache-2.0
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
#ifndef ASR_WAV2LETTER_PREPROCESS_HPP
#define ASR_WAV2LETTER_PREPROCESS_HPP
#include "TensorFlowLiteMicro.hpp"
#include "Wav2LetterMfcc.hpp"
#include "AudioUtils.hpp"
#include "DataStructures.hpp"
#include "BaseProcessing.hpp"
#include "log_macros.h"
namespace arm {
namespace app {
/* Class to facilitate pre-processing calculation for Wav2Letter model
* for ASR. */
using AudioWindow = audio::SlidingWindow<const int16_t>;
class AsrPreProcess : public BasePreProcess {
public:
/**
* @brief Constructor.
* @param[in] inputTensor Pointer to the TFLite Micro input Tensor.
* @param[in] numMfccFeatures Number of MFCC features per window.
* @param[in] numFeatureFrames Number of MFCC vectors that need to be calculated
* for an inference.
* @param[in] mfccWindowLen Number of audio elements to calculate MFCC features per window.
* @param[in] mfccWindowStride Stride (in number of elements) for moving the MFCC window.
*/
AsrPreProcess(TfLiteTensor* inputTensor,
uint32_t numMfccFeatures,
uint32_t numFeatureFrames,
uint32_t mfccWindowLen,
uint32_t mfccWindowStride);
/**
* @brief Calculates the features required from audio data. This
* includes MFCC, first and second order deltas,
* normalisation and finally, quantisation. The tensor is
* populated with features from a given window placed along
* in a single row.
* @param[in] audioData Pointer to the first element of audio data.
* @param[in] audioDataLen Number of elements in the audio data.
* @return true if successful, false in case of error.
*/
bool DoPreProcess(const void* audioData, size_t audioDataLen) override;
protected:
/**
* @brief Computes the first and second order deltas for the
* MFCC buffers - they are assumed to be populated.
*
* @param[in] mfcc MFCC buffers.
* @param[out] delta1 Result of the first diff computation.
* @param[out] delta2 Result of the second diff computation.
* @return true if successful, false otherwise.
*/
static bool ComputeDeltas(Array2d<float>& mfcc,
Array2d<float>& delta1,
Array2d<float>& delta2);
/**
* @brief Given a 2D vector of floats, rescale it to have mean of 0 and
* standard deviation of 1.
* @param[in,out] vec Vector of vector of floats.
*/
static void StandardizeVecF32(Array2d<float>& vec);
/**
* @brief Standardizes all the MFCC and delta buffers to have mean 0 and std. dev 1.
*/
void Standarize();
/**
* @brief Given the quantisation and data type limits, computes
* the quantised values of a floating point input data.
* @param[in] elem Element to be quantised.
* @param[in] quantScale Scale.
* @param[in] quantOffset Offset.
* @param[in] minVal Numerical limit - minimum.
* @param[in] maxVal Numerical limit - maximum.
* @return Floating point quantised value.
*/
static float GetQuantElem(
float elem,
float quantScale,
int quantOffset,
float minVal,
float maxVal);
/**
* @brief Quantises the MFCC and delta buffers, and places them
* in the output buffer. While doing so, it transposes
* the data. Reason: Buffers in this class are arranged
* for "time" axis to be row major. Primary reason for
* this being the convolution speed up (as we can use
* contiguous memory). The output, however, requires the
* time axis to be in column major arrangement.
* @param[in] outputBuf Pointer to the output buffer.
* @param[in] outputBufSz Output buffer's size.
* @param[in] quantScale Quantisation scale.
* @param[in] quantOffset Quantisation offset.
*/
template <typename T>
bool Quantise(
T* outputBuf,
const uint32_t outputBufSz,
const float quantScale,
const int quantOffset)
{
/* Check the output size will fit everything. */
if (outputBufSz < (this->m_mfccBuf.size(0) * 3 * sizeof(T))) {
printf_err("Tensor size too small for features\n");
return false;
}
/* Populate. */
T* outputBufMfcc = outputBuf;
T* outputBufD1 = outputBuf + this->m_numMfccFeats;
T* outputBufD2 = outputBufD1 + this->m_numMfccFeats;
const uint32_t ptrIncr = this->m_numMfccFeats * 2; /* (3 vectors - 1 vector) */
const float minVal = std::numeric_limits<T>::min();
const float maxVal = std::numeric_limits<T>::max();
/* Need to transpose while copying and concatenating the tensor. */
for (uint32_t j = 0; j < this->m_numFeatureFrames; ++j) {
for (uint32_t i = 0; i < this->m_numMfccFeats; ++i) {
*outputBufMfcc++ = static_cast<T>(AsrPreProcess::GetQuantElem(
this->m_mfccBuf(i, j), quantScale,
quantOffset, minVal, maxVal));
*outputBufD1++ = static_cast<T>(AsrPreProcess::GetQuantElem(
this->m_delta1Buf(i, j), quantScale,
quantOffset, minVal, maxVal));
*outputBufD2++ = static_cast<T>(AsrPreProcess::GetQuantElem(
this->m_delta2Buf(i, j), quantScale,
quantOffset, minVal, maxVal));
}
outputBufMfcc += ptrIncr;
outputBufD1 += ptrIncr;
outputBufD2 += ptrIncr;
}
return true;
}
private:
audio::Wav2LetterMFCC m_mfcc; /* MFCC instance. */
TfLiteTensor* m_inputTensor; /* Model input tensor. */
/* Actual buffers to be populated. */
Array2d<float> m_mfccBuf; /* Contiguous buffer 1D: MFCC */
Array2d<float> m_delta1Buf; /* Contiguous buffer 1D: Delta 1 */
Array2d<float> m_delta2Buf; /* Contiguous buffer 1D: Delta 2 */
uint32_t m_mfccWindowLen; /* Window length for MFCC. */
uint32_t m_mfccWindowStride; /* Window stride len for MFCC. */
uint32_t m_numMfccFeats; /* Number of MFCC features per window. */
uint32_t m_numFeatureFrames; /* How many sets of m_numMfccFeats. */
AudioWindow m_mfccSlidingWindow; /* Sliding window to calculate MFCCs. */
};
} /* namespace app */
} /* namespace arm */
#endif /* ASR_WAV2LETTER_PREPROCESS_HPP */
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