/* * Copyright (c) 2021 Arm Limited. All rights reserved. * 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. */ #include "UseCaseHandler.hpp" #include "hal.h" #include "InputFiles.hpp" #include "AudioUtils.hpp" #include "UseCaseCommonUtils.hpp" #include "DsCnnModel.hpp" #include "DsCnnMfcc.hpp" #include "Classifier.hpp" #include "KwsResult.hpp" #include "Wav2LetterMfcc.hpp" #include "Wav2LetterPreprocess.hpp" #include "Wav2LetterPostprocess.hpp" #include "AsrResult.hpp" #include "AsrClassifier.hpp" #include "OutputDecode.hpp" using KwsClassifier = arm::app::Classifier; namespace arm { namespace app { enum AsrOutputReductionAxis { AxisRow = 1, AxisCol = 2 }; struct KWSOutput { bool executionSuccess = false; const int16_t* asrAudioStart = nullptr; int32_t asrAudioSamples = 0; }; /** * @brief Helper function to increment current audio clip index * @param[in,out] ctx pointer to the application context object **/ static void IncrementAppCtxClipIdx(ApplicationContext& ctx); /** * @brief Helper function to set the audio clip index * @param[in,out] ctx pointer to the application context object * @param[in] idx value to be set * @return true if index is set, false otherwise **/ static bool SetAppCtxClipIdx(ApplicationContext& ctx, uint32_t idx); /** * @brief Presents kws inference results using the data presentation * object. * @param[in] platform reference to the hal platform object * @param[in] results vector of classification results to be displayed * @return true if successful, false otherwise **/ static bool PresentInferenceResult(hal_platform& platform, std::vector& results); /** * @brief Presents asr inference results using the data presentation * object. * @param[in] platform reference to the hal platform object * @param[in] results vector of classification results to be displayed * @return true if successful, false otherwise **/ static bool PresentInferenceResult(hal_platform& platform, std::vector& results); /** * @brief Returns a function to perform feature calculation and populates input tensor data with * MFCC data. * * Input tensor data type check is performed to choose correct MFCC feature data type. * If tensor has an integer data type then original features are quantised. * * Warning: mfcc calculator provided as input must have the same life scope as returned function. * * @param[in] mfcc MFCC feature calculator. * @param[in,out] inputTensor Input tensor pointer to store calculated features. * @param[in] cacheSize Size of the feture vectors cache (number of feature vectors). * * @return function function to be called providing audio sample and sliding window index. **/ static std::function&, int, bool, size_t)> GetFeatureCalculator(audio::DsCnnMFCC& mfcc, TfLiteTensor* inputTensor, size_t cacheSize); /** * @brief Performs the KWS pipeline. * @param[in,out] ctx pointer to the application context object * * @return KWSOutput struct containing pointer to audio data where ASR should begin * and how much data to process. */ static KWSOutput doKws(ApplicationContext& ctx) { constexpr uint32_t dataPsnTxtInfStartX = 20; constexpr uint32_t dataPsnTxtInfStartY = 40; constexpr int minTensorDims = static_cast( (arm::app::DsCnnModel::ms_inputRowsIdx > arm::app::DsCnnModel::ms_inputColsIdx)? arm::app::DsCnnModel::ms_inputRowsIdx : arm::app::DsCnnModel::ms_inputColsIdx); KWSOutput output; auto& profiler = ctx.Get("profiler"); auto& kwsModel = ctx.Get("kwsmodel"); if (!kwsModel.IsInited()) { printf_err("KWS model has not been initialised\n"); return output; } const int kwsFrameLength = ctx.Get("kwsframeLength"); const int kwsFrameStride = ctx.Get("kwsframeStride"); const float kwsScoreThreshold = ctx.Get("kwsscoreThreshold"); TfLiteTensor* kwsOutputTensor = kwsModel.GetOutputTensor(0); TfLiteTensor* kwsInputTensor = kwsModel.GetInputTensor(0); if (!kwsInputTensor->dims) { printf_err("Invalid input tensor dims\n"); return output; } else if (kwsInputTensor->dims->size < minTensorDims) { printf_err("Input tensor dimension should be >= %d\n", minTensorDims); return output; } const uint32_t kwsNumMfccFeats = ctx.Get("kwsNumMfcc"); const uint32_t kwsNumAudioWindows = ctx.Get("kwsNumAudioWins"); audio::DsCnnMFCC kwsMfcc = audio::DsCnnMFCC(kwsNumMfccFeats, kwsFrameLength); kwsMfcc.Init(); /* Deduce the data length required for 1 KWS inference from the network parameters. */ auto kwsAudioDataWindowSize = kwsNumAudioWindows * kwsFrameStride + (kwsFrameLength - kwsFrameStride); auto kwsMfccWindowSize = kwsFrameLength; auto kwsMfccWindowStride = kwsFrameStride; /* We are choosing to move by half the window size => for a 1 second window size, * this means an overlap of 0.5 seconds. */ auto kwsAudioDataStride = kwsAudioDataWindowSize / 2; info("KWS audio data window size %" PRIu32 "\n", kwsAudioDataWindowSize); /* Stride must be multiple of mfcc features window stride to re-use features. */ if (0 != kwsAudioDataStride % kwsMfccWindowStride) { kwsAudioDataStride -= kwsAudioDataStride % kwsMfccWindowStride; } auto kwsMfccVectorsInAudioStride = kwsAudioDataStride/kwsMfccWindowStride; /* We expect to be sampling 1 second worth of data at a time * NOTE: This is only used for time stamp calculation. */ const float kwsAudioParamsSecondsPerSample = 1.0/audio::DsCnnMFCC::ms_defaultSamplingFreq; auto currentIndex = ctx.Get("clipIndex"); /* Creating a mfcc features sliding window for the data required for 1 inference. */ auto kwsAudioMFCCWindowSlider = audio::SlidingWindow( get_audio_array(currentIndex), kwsAudioDataWindowSize, kwsMfccWindowSize, kwsMfccWindowStride); /* Creating a sliding window through the whole audio clip. */ auto audioDataSlider = audio::SlidingWindow( get_audio_array(currentIndex), get_audio_array_size(currentIndex), kwsAudioDataWindowSize, kwsAudioDataStride); /* Calculate number of the feature vectors in the window overlap region. * These feature vectors will be reused.*/ size_t numberOfReusedFeatureVectors = kwsAudioMFCCWindowSlider.TotalStrides() + 1 - kwsMfccVectorsInAudioStride; auto kwsMfccFeatureCalc = GetFeatureCalculator(kwsMfcc, kwsInputTensor, numberOfReusedFeatureVectors); if (!kwsMfccFeatureCalc){ return output; } /* Container for KWS results. */ std::vector kwsResults; /* Display message on the LCD - inference running. */ auto& platform = ctx.Get("platform"); std::string str_inf{"Running KWS inference... "}; platform.data_psn->present_data_text( str_inf.c_str(), str_inf.size(), dataPsnTxtInfStartX, dataPsnTxtInfStartY, false); info("Running KWS inference on audio clip %" PRIu32 " => %s\n", currentIndex, get_filename(currentIndex)); /* Start sliding through audio clip. */ while (audioDataSlider.HasNext()) { const int16_t* inferenceWindow = audioDataSlider.Next(); /* We moved to the next window - set the features sliding to the new address. */ kwsAudioMFCCWindowSlider.Reset(inferenceWindow); /* The first window does not have cache ready. */ bool useCache = audioDataSlider.Index() > 0 && numberOfReusedFeatureVectors > 0; /* Start calculating features inside one audio sliding window. */ while (kwsAudioMFCCWindowSlider.HasNext()) { const int16_t* kwsMfccWindow = kwsAudioMFCCWindowSlider.Next(); std::vector kwsMfccAudioData = std::vector(kwsMfccWindow, kwsMfccWindow + kwsMfccWindowSize); /* Compute features for this window and write them to input tensor. */ kwsMfccFeatureCalc(kwsMfccAudioData, kwsAudioMFCCWindowSlider.Index(), useCache, kwsMfccVectorsInAudioStride); } info("Inference %zu/%zu\n", audioDataSlider.Index() + 1, audioDataSlider.TotalStrides() + 1); /* Run inference over this audio clip sliding window. */ if (!RunInference(kwsModel, profiler)) { printf_err("KWS inference failed\n"); return output; } std::vector kwsClassificationResult; auto& kwsClassifier = ctx.Get("kwsclassifier"); kwsClassifier.GetClassificationResults( kwsOutputTensor, kwsClassificationResult, ctx.Get&>("kwslabels"), 1); kwsResults.emplace_back( kws::KwsResult( kwsClassificationResult, audioDataSlider.Index() * kwsAudioParamsSecondsPerSample * kwsAudioDataStride, audioDataSlider.Index(), kwsScoreThreshold) ); /* Keyword detected. */ if (kwsClassificationResult[0].m_labelIdx == ctx.Get("keywordindex")) { output.asrAudioStart = inferenceWindow + kwsAudioDataWindowSize; output.asrAudioSamples = get_audio_array_size(currentIndex) - (audioDataSlider.NextWindowStartIndex() - kwsAudioDataStride + kwsAudioDataWindowSize); break; } #if VERIFY_TEST_OUTPUT arm::app::DumpTensor(kwsOutputTensor); #endif /* VERIFY_TEST_OUTPUT */ } /* while (audioDataSlider.HasNext()) */ /* Erase. */ str_inf = std::string(str_inf.size(), ' '); platform.data_psn->present_data_text( str_inf.c_str(), str_inf.size(), dataPsnTxtInfStartX, dataPsnTxtInfStartY, false); if (!PresentInferenceResult(platform, kwsResults)) { return output; } profiler.PrintProfilingResult(); output.executionSuccess = true; return output; } /** * @brief Performs the ASR pipeline. * * @param[in,out] ctx pointer to the application context object * @param[in] kwsOutput struct containing pointer to audio data where ASR should begin * and how much data to process * @return bool true if pipeline executed without failure */ static bool doAsr(ApplicationContext& ctx, const KWSOutput& kwsOutput) { constexpr uint32_t dataPsnTxtInfStartX = 20; constexpr uint32_t dataPsnTxtInfStartY = 40; auto& profiler = ctx.Get("profiler"); auto& platform = ctx.Get("platform"); platform.data_psn->clear(COLOR_BLACK); /* Get model reference. */ auto& asrModel = ctx.Get("asrmodel"); if (!asrModel.IsInited()) { printf_err("ASR model has not been initialised\n"); return false; } /* Get score threshold to be applied for the classifier (post-inference). */ auto asrScoreThreshold = ctx.Get("asrscoreThreshold"); /* Dimensions of the tensor should have been verified by the callee. */ TfLiteTensor* asrInputTensor = asrModel.GetInputTensor(0); TfLiteTensor* asrOutputTensor = asrModel.GetOutputTensor(0); const uint32_t asrInputRows = asrInputTensor->dims->data[arm::app::Wav2LetterModel::ms_inputRowsIdx]; /* Populate ASR MFCC related parameters. */ auto asrMfccParamsWinLen = ctx.Get("asrframeLength"); auto asrMfccParamsWinStride = ctx.Get("asrframeStride"); /* Populate ASR inference context and inner lengths for input. */ auto asrInputCtxLen = ctx.Get("ctxLen"); const uint32_t asrInputInnerLen = asrInputRows - (2 * asrInputCtxLen); /* Make sure the input tensor supports the above context and inner lengths. */ if (asrInputRows <= 2 * asrInputCtxLen || asrInputRows <= asrInputInnerLen) { printf_err("ASR input rows not compatible with ctx length %" PRIu32 "\n", asrInputCtxLen); return false; } /* Audio data stride corresponds to inputInnerLen feature vectors. */ const uint32_t asrAudioParamsWinLen = (asrInputRows - 1) * asrMfccParamsWinStride + (asrMfccParamsWinLen); const uint32_t asrAudioParamsWinStride = asrInputInnerLen * asrMfccParamsWinStride; const float asrAudioParamsSecondsPerSample = (1.0/audio::Wav2LetterMFCC::ms_defaultSamplingFreq); /* Get pre/post-processing objects */ auto& asrPrep = ctx.Get("preprocess"); auto& asrPostp = ctx.Get("postprocess"); /* Set default reduction axis for post-processing. */ const uint32_t reductionAxis = arm::app::Wav2LetterModel::ms_outputRowsIdx; /* Get the remaining audio buffer and respective size from KWS results. */ const int16_t* audioArr = kwsOutput.asrAudioStart; const uint32_t audioArrSize = kwsOutput.asrAudioSamples; /* Audio clip must have enough samples to produce 1 MFCC feature. */ std::vector audioBuffer = std::vector(audioArr, audioArr + audioArrSize); if (audioArrSize < asrMfccParamsWinLen) { printf_err("Not enough audio samples, minimum needed is %" PRIu32 "\n", asrMfccParamsWinLen); return false; } /* Initialise an audio slider. */ auto audioDataSlider = audio::FractionalSlidingWindow( audioBuffer.data(), audioBuffer.size(), asrAudioParamsWinLen, asrAudioParamsWinStride); /* Declare a container for results. */ std::vector asrResults; /* Display message on the LCD - inference running. */ std::string str_inf{"Running ASR inference... "}; platform.data_psn->present_data_text( str_inf.c_str(), str_inf.size(), dataPsnTxtInfStartX, dataPsnTxtInfStartY, false); size_t asrInferenceWindowLen = asrAudioParamsWinLen; /* Start sliding through audio clip. */ while (audioDataSlider.HasNext()) { /* If not enough audio see how much can be sent for processing. */ size_t nextStartIndex = audioDataSlider.NextWindowStartIndex(); if (nextStartIndex + asrAudioParamsWinLen > audioBuffer.size()) { asrInferenceWindowLen = audioBuffer.size() - nextStartIndex; } const int16_t* asrInferenceWindow = audioDataSlider.Next(); info("Inference %zu/%zu\n", audioDataSlider.Index() + 1, static_cast(ceilf(audioDataSlider.FractionalTotalStrides() + 1))); /* Calculate MFCCs, deltas and populate the input tensor. */ asrPrep.Invoke(asrInferenceWindow, asrInferenceWindowLen, asrInputTensor); /* Run inference over this audio clip sliding window. */ if (!RunInference(asrModel, profiler)) { printf_err("ASR inference failed\n"); return false; } /* Post-process. */ asrPostp.Invoke(asrOutputTensor, reductionAxis, !audioDataSlider.HasNext()); /* Get results. */ std::vector asrClassificationResult; auto& asrClassifier = ctx.Get("asrclassifier"); asrClassifier.GetClassificationResults( asrOutputTensor, asrClassificationResult, ctx.Get&>("asrlabels"), 1); asrResults.emplace_back(asr::AsrResult(asrClassificationResult, (audioDataSlider.Index() * asrAudioParamsSecondsPerSample * asrAudioParamsWinStride), audioDataSlider.Index(), asrScoreThreshold)); #if VERIFY_TEST_OUTPUT arm::app::DumpTensor(asrOutputTensor, asrOutputTensor->dims->data[arm::app::Wav2LetterModel::ms_outputColsIdx]); #endif /* VERIFY_TEST_OUTPUT */ /* Erase */ str_inf = std::string(str_inf.size(), ' '); platform.data_psn->present_data_text( str_inf.c_str(), str_inf.size(), dataPsnTxtInfStartX, dataPsnTxtInfStartY, false); } if (!PresentInferenceResult(platform, asrResults)) { return false; } profiler.PrintProfilingResult(); return true; } /* Audio inference classification handler. */ bool ClassifyAudioHandler(ApplicationContext& ctx, uint32_t clipIndex, bool runAll) { auto& platform = ctx.Get("platform"); platform.data_psn->clear(COLOR_BLACK); /* If the request has a valid size, set the audio index. */ if (clipIndex < NUMBER_OF_FILES) { if (!SetAppCtxClipIdx(ctx, clipIndex)) { return false; } } auto startClipIdx = ctx.Get("clipIndex"); do { KWSOutput kwsOutput = doKws(ctx); if (!kwsOutput.executionSuccess) { return false; } if (kwsOutput.asrAudioStart != nullptr && kwsOutput.asrAudioSamples > 0) { info("Keyword spotted\n"); if(!doAsr(ctx, kwsOutput)) { printf_err("ASR failed"); return false; } } IncrementAppCtxClipIdx(ctx); } while (runAll && ctx.Get("clipIndex") != startClipIdx); return true; } static void IncrementAppCtxClipIdx(ApplicationContext& ctx) { auto curAudioIdx = ctx.Get("clipIndex"); if (curAudioIdx + 1 >= NUMBER_OF_FILES) { ctx.Set("clipIndex", 0); return; } ++curAudioIdx; ctx.Set("clipIndex", curAudioIdx); } static bool SetAppCtxClipIdx(ApplicationContext& ctx, uint32_t idx) { if (idx >= NUMBER_OF_FILES) { printf_err("Invalid idx %" PRIu32 " (expected less than %u)\n", idx, NUMBER_OF_FILES); return false; } ctx.Set("clipIndex", idx); return true; } static bool PresentInferenceResult(hal_platform& platform, std::vector& results) { constexpr uint32_t dataPsnTxtStartX1 = 20; constexpr uint32_t dataPsnTxtStartY1 = 30; constexpr uint32_t dataPsnTxtYIncr = 16; /* Row index increment. */ platform.data_psn->set_text_color(COLOR_GREEN); /* Display each result. */ uint32_t rowIdx1 = dataPsnTxtStartY1 + 2 * dataPsnTxtYIncr; for (uint32_t i = 0; i < results.size(); ++i) { std::string topKeyword{""}; float score = 0.f; if (!results[i].m_resultVec.empty()) { topKeyword = results[i].m_resultVec[0].m_label; score = results[i].m_resultVec[0].m_normalisedVal; } std::string resultStr = std::string{"@"} + std::to_string(results[i].m_timeStamp) + std::string{"s: "} + topKeyword + std::string{" ("} + std::to_string(static_cast(score * 100)) + std::string{"%)"}; platform.data_psn->present_data_text( resultStr.c_str(), resultStr.size(), dataPsnTxtStartX1, rowIdx1, 0); rowIdx1 += dataPsnTxtYIncr; info("For timestamp: %f (inference #: %" PRIu32 "); threshold: %f\n", results[i].m_timeStamp, results[i].m_inferenceNumber, results[i].m_threshold); for (uint32_t j = 0; j < results[i].m_resultVec.size(); ++j) { info("\t\tlabel @ %" PRIu32 ": %s, score: %f\n", j, results[i].m_resultVec[j].m_label.c_str(), results[i].m_resultVec[j].m_normalisedVal); } } return true; } static bool PresentInferenceResult(hal_platform& platform, std::vector& results) { constexpr uint32_t dataPsnTxtStartX1 = 20; constexpr uint32_t dataPsnTxtStartY1 = 80; constexpr bool allow_multiple_lines = true; platform.data_psn->set_text_color(COLOR_GREEN); /* Results from multiple inferences should be combined before processing. */ std::vector combinedResults; for (auto& result : results) { combinedResults.insert(combinedResults.end(), result.m_resultVec.begin(), result.m_resultVec.end()); } for (auto& result : results) { /* Get the final result string using the decoder. */ std::string infResultStr = audio::asr::DecodeOutput(result.m_resultVec); info("Result for inf %" PRIu32 ": %s\n", result.m_inferenceNumber, infResultStr.c_str()); } std::string finalResultStr = audio::asr::DecodeOutput(combinedResults); platform.data_psn->present_data_text( finalResultStr.c_str(), finalResultStr.size(), dataPsnTxtStartX1, dataPsnTxtStartY1, allow_multiple_lines); info("Final result: %s\n", finalResultStr.c_str()); return true; } /** * @brief Generic feature calculator factory. * * Returns lambda function to compute features using features cache. * Real features math is done by a lambda function provided as a parameter. * Features are written to input tensor memory. * * @tparam T feature vector type. * @param inputTensor model input tensor pointer. * @param cacheSize number of feature vectors to cache. Defined by the sliding window overlap. * @param compute features calculator function. * @return lambda function to compute features. **/ template std::function&, size_t, bool, size_t)> FeatureCalc(TfLiteTensor* inputTensor, size_t cacheSize, std::function (std::vector& )> compute) { /* Feature cache to be captured by lambda function. */ static std::vector> featureCache = std::vector>(cacheSize); return [=](std::vector& audioDataWindow, size_t index, bool useCache, size_t featuresOverlapIndex) { T* tensorData = tflite::GetTensorData(inputTensor); std::vector features; /* Reuse features from cache if cache is ready and sliding windows overlap. * Overlap is in the beginning of sliding window with a size of a feature cache. */ if (useCache && index < featureCache.size()) { features = std::move(featureCache[index]); } else { features = std::move(compute(audioDataWindow)); } auto size = features.size(); auto sizeBytes = sizeof(T) * size; std::memcpy(tensorData + (index * size), features.data(), sizeBytes); /* Start renewing cache as soon iteration goes out of the windows overlap. */ if (index >= featuresOverlapIndex) { featureCache[index - featuresOverlapIndex] = std::move(features); } }; } template std::function&, size_t , bool, size_t)> FeatureCalc(TfLiteTensor* inputTensor, size_t cacheSize, std::function (std::vector& )> compute); template std::function&, size_t , bool, size_t)> FeatureCalc(TfLiteTensor* inputTensor, size_t cacheSize, std::function (std::vector& )> compute); template std::function&, size_t , bool, size_t)> FeatureCalc(TfLiteTensor* inputTensor, size_t cacheSize, std::function (std::vector& )> compute); template std::function&, size_t, bool, size_t)> FeatureCalc(TfLiteTensor* inputTensor, size_t cacheSize, std::function(std::vector&)> compute); static std::function&, int, bool, size_t)> GetFeatureCalculator(audio::DsCnnMFCC& mfcc, TfLiteTensor* inputTensor, size_t cacheSize) { std::function&, size_t, bool, size_t)> mfccFeatureCalc; TfLiteQuantization quant = inputTensor->quantization; if (kTfLiteAffineQuantization == quant.type) { auto* quantParams = (TfLiteAffineQuantization*) quant.params; const float quantScale = quantParams->scale->data[0]; const int quantOffset = quantParams->zero_point->data[0]; switch (inputTensor->type) { case kTfLiteInt8: { mfccFeatureCalc = FeatureCalc(inputTensor, cacheSize, [=, &mfcc](std::vector& audioDataWindow) { return mfcc.MfccComputeQuant(audioDataWindow, quantScale, quantOffset); } ); break; } case kTfLiteUInt8: { mfccFeatureCalc = FeatureCalc(inputTensor, cacheSize, [=, &mfcc](std::vector& audioDataWindow) { return mfcc.MfccComputeQuant(audioDataWindow, quantScale, quantOffset); } ); break; } case kTfLiteInt16: { mfccFeatureCalc = FeatureCalc(inputTensor, cacheSize, [=, &mfcc](std::vector& audioDataWindow) { return mfcc.MfccComputeQuant(audioDataWindow, quantScale, quantOffset); } ); break; } default: printf_err("Tensor type %s not supported\n", TfLiteTypeGetName(inputTensor->type)); } } else { mfccFeatureCalc = mfccFeatureCalc = FeatureCalc(inputTensor, cacheSize, [&mfcc](std::vector& audioDataWindow) { return mfcc.MfccCompute(audioDataWindow); }); } return mfccFeatureCalc; } } /* namespace app */ } /* namespace arm */