#include #include #include #include #include #include #include #include #include #include #include #include using namespace cv; using namespace std; class FilterbankFeatures { // Initializes pre-processing class. Default values are the values used by the Jasper // architecture for pre-processing. For more details, refer to the paper here: // https://arxiv.org/abs/1904.03288 private: int sample_rate = 16000; double window_size = 0.02; double window_stride = 0.01; int win_length = static_cast(sample_rate * window_size); // Number of samples in window int hop_length = static_cast(sample_rate * window_stride); // Number of steps to advance between frames int n_fft = 512; // Size of window for STFT // Parameters for filterbanks calculation int n_filt = 64; double lowfreq = 0.; double highfreq = sample_rate / 2; public: // Mel filterbanks preperation double hz_to_mel(double frequencies) { //Converts frequencies from hz to mel scale // Fill in the linear scale double f_min = 0.0; double f_sp = 200.0 / 3; double mels = (frequencies - f_min) / f_sp; // Fill in the log-scale part double min_log_hz = 1000.0; // beginning of log region (Hz) double min_log_mel = (min_log_hz - f_min) / f_sp; // same (Mels) double logstep = std::log(6.4) / 27.0; // step size for log region if (frequencies >= min_log_hz) { mels = min_log_mel + std::log(frequencies / min_log_hz) / logstep; } return mels; } vector mel_to_hz(vector& mels) { // Converts frequencies from mel to hz scale // Fill in the linear scale double f_min = 0.0; double f_sp = 200.0 / 3; vector freqs; for (size_t i = 0; i < mels.size(); i++) { freqs.push_back(f_min + f_sp * mels[i]); } // And now the nonlinear scale double min_log_hz = 1000.0; // beginning of log region (Hz) double min_log_mel = (min_log_hz - f_min) / f_sp; // same (Mels) double logstep = std::log(6.4) / 27.0; // step size for log region for(size_t i = 0; i < mels.size(); i++) { if (mels[i] >= min_log_mel) { freqs[i] = min_log_hz * exp(logstep * (mels[i] - min_log_mel)); } } return freqs; } vector mel_frequencies(int n_mels, double fmin, double fmax) { // Calculates n mel frequencies between 2 frequencies double min_mel = hz_to_mel(fmin); double max_mel = hz_to_mel(fmax); vector mels; double step = (max_mel - min_mel) / (n_mels - 1); for(double i = min_mel; i < max_mel; i += step) { mels.push_back(i); } mels.push_back(max_mel); vector res = mel_to_hz(mels); return res; } vector> mel(int n_mels, double fmin, double fmax) { // Generates mel filterbank matrix double num = 1 + n_fft / 2; vector> weights(n_mels, vector(static_cast(num), 0.)); // Center freqs of each FFT bin vector fftfreqs; double step = (sample_rate / 2) / (num - 1); for(double i = 0; i <= sample_rate / 2; i += step) { fftfreqs.push_back(i); } // 'Center freqs' of mel bands - uniformly spaced between limits vector mel_f = mel_frequencies(n_mels + 2, fmin, fmax); vector fdiff; for(size_t i = 1; i < mel_f.size(); ++i) { fdiff.push_back(mel_f[i]- mel_f[i - 1]); } vector> ramps(mel_f.size(), vector(fftfreqs.size())); for (size_t i = 0; i < mel_f.size(); ++i) { for (size_t j = 0; j < fftfreqs.size(); ++j) { ramps[i][j] = mel_f[i] - fftfreqs[j]; } } double lower, upper, enorm; for (int i = 0; i < n_mels; ++i) { // using Slaney-style mel which is scaled to be approx constant energy per channel enorm = 2./(mel_f[i + 2] - mel_f[i]); for (int j = 0; j < static_cast(num); ++j) { // lower and upper slopes for all bins lower = (-1) * ramps[i][j] / fdiff[i]; upper = ramps[i + 2][j] / fdiff[i + 1]; weights[i][j] = max(0., min(lower, upper)) * enorm; } } return weights; } // STFT preperation vector pad_window_center(vector&data, int size) { // Pad the window out to n_fft size int n = static_cast(data.size()); int lpad = static_cast((size - n) / 2); vector pad_array; for(int i = 0; i < lpad; ++i) { pad_array.push_back(0.); } for(size_t i = 0; i < data.size(); ++i) { pad_array.push_back(data[i]); } for(int i = 0; i < lpad; ++i) { pad_array.push_back(0.); } return pad_array; } vector> frame(vector& x) { // Slices a data array into overlapping frames. int n_frames = static_cast(1 + (x.size() - n_fft) / hop_length); vector> new_x(n_fft, vector(n_frames)); for (int i = 0; i < n_fft; ++i) { for (int j = 0; j < n_frames; ++j) { new_x[i][j] = x[i + j * hop_length]; } } return new_x; } vector hanning() { // https://en.wikipedia.org/wiki/Window_function#Hann_and_Hamming_windows vector window_tensor; for (int j = 1 - win_length; j < win_length; j+=2) { window_tensor.push_back(1 - (0.5 * (1 - cos(CV_PI * j / (win_length - 1))))); } return window_tensor; } vector> stft_power(vector& y) { // Short Time Fourier Transform. The STFT represents a signal in the time-frequency // domain by computing discrete Fourier transforms (DFT) over short overlapping windows. // https://en.wikipedia.org/wiki/Short-time_Fourier_transform // Pad the time series so that frames are centered vector new_y; int num = int(n_fft / 2); for (int i = 0; i < num; ++i) { new_y.push_back(y[num - i]); } for (size_t i = 0; i < y.size(); ++i) { new_y.push_back(y[i]); } for (size_t i = y.size() - 2; i >= y.size() - num - 1; --i) { new_y.push_back(y[i]); } // Compute a window function vector window_tensor = hanning(); // Pad the window out to n_fft size vector fft_window = pad_window_center(window_tensor, n_fft); // Window the time series vector> y_frames = frame(new_y); // Multiply on fft_window for (size_t i = 0; i < y_frames.size(); ++i) { for (size_t j = 0; j < y_frames[0].size(); ++j) { y_frames[i][j] *= fft_window[i]; } } // Transpose frames for computing stft vector> y_frames_transpose(y_frames[0].size(), vector(y_frames.size())); for (size_t i = 0; i < y_frames[0].size(); ++i) { for (size_t j = 0; j < y_frames.size(); ++j) { y_frames_transpose[i][j] = y_frames[j][i]; } } // Short Time Fourier Transform // and get power of spectrum vector> spectrum_power(y_frames_transpose[0].size() / 2 + 1 ); for (size_t i = 0; i < y_frames_transpose.size(); ++i) { Mat dstMat; dft(y_frames_transpose[i], dstMat, DFT_COMPLEX_OUTPUT); // we need only the first part of the spectrum, the second part is symmetrical for (int j = 0; j < static_cast(y_frames_transpose[0].size()) / 2 + 1; ++j) { double power_re = dstMat.at(2 * j) * dstMat.at(2 * j); double power_im = dstMat.at(2 * j + 1) * dstMat.at(2 * j + 1); spectrum_power[j].push_back(power_re + power_im); } } return spectrum_power; } Mat calculate_features(vector& x) { // Calculates filterbank features matrix. // Do preemphasis std::default_random_engine generator; std::normal_distribution normal_distr(0, 1); double dither = 1e-5; for(size_t i = 0; i < x.size(); ++i) { x[i] += dither * static_cast(normal_distr(generator)); } double preemph = 0.97; for (size_t i = x.size() - 1; i > 0; --i) { x[i] -= preemph * x[i-1]; } // Calculate Short Time Fourier Transform and get power of spectrum auto spectrum_power = stft_power(x); vector> filterbanks = mel(n_filt, lowfreq, highfreq); // Calculate log of multiplication of filterbanks matrix on spectrum_power matrix vector> x_stft(filterbanks.size(), vector(spectrum_power[0].size(), 0)); for (size_t i = 0; i < filterbanks.size(); ++i) { for (size_t j = 0; j < filterbanks[0].size(); ++j) { for (size_t k = 0; k < spectrum_power[0].size(); ++k) { x_stft[i][k] += filterbanks[i][j] * spectrum_power[j][k]; } } for (size_t k = 0; k < spectrum_power[0].size(); ++k) { x_stft[i][k] = std::log(x_stft[i][k] + 1e-20); } } // normalize data auto elments_num = x_stft[0].size(); for(size_t i = 0; i < x_stft.size(); ++i) { double x_mean = std::accumulate(x_stft[i].begin(), x_stft[i].end(), 0.) / elments_num; // arithmetic mean double x_std = 0; // standard deviation for(size_t j = 0; j < elments_num; ++j) { double subtract = x_stft[i][j] - x_mean; x_std += subtract * subtract; } x_std /= elments_num; x_std = sqrt(x_std) + 1e-10; // make sure x_std is not zero for(size_t j = 0; j < elments_num; ++j) { x_stft[i][j] = (x_stft[i][j] - x_mean) / x_std; // standard score } } Mat calculate_features(static_cast(x_stft.size()), static_cast(x_stft[0].size()), CV_32F); for(int i = 0; i < calculate_features.size[0]; ++i) { for(int j = 0; j < calculate_features.size[1]; ++j) { calculate_features.at(i, j) = static_cast(x_stft[i][j]); } } return calculate_features; } }; class Decoder { // Used for decoding the output of jasper model private: unordered_map labels_map = fillMap(); int blank_id = 28; public: unordered_map fillMap() { vector labels={' ','a','b','c','d','e','f','g','h','i','j','k','l','m','n','o','p' ,'q','r','s','t','u','v','w','x','y','z','\''}; unordered_map map; for(int i = 0; i < static_cast(labels.size()); ++i) { map[i] = labels[i]; } return map; } string decode(Mat& x) { // Takes output of Jasper model and performs ctc decoding algorithm to // remove duplicates and special symbol. Returns prediction vector prediction; for(int i = 0; i < x.size[1]; ++i) { double maxEl = -1e10; int ind = 0; for(int j = 0; j < x.size[2]; ++j) { if (maxEl <= x.at(0, i, j)) { maxEl = x.at(0, i, j); ind = j; } } prediction.push_back(ind); } // CTC decoding procedure vector decoded_prediction = {}; int previous = blank_id; for(int i = 0; i < static_cast(prediction.size()); ++i) { if (( prediction[i] != previous || previous == blank_id) && prediction[i] != blank_id) { decoded_prediction.push_back(prediction[i]); } previous = prediction[i]; } string hypotheses = {}; for(size_t i = 0; i < decoded_prediction.size(); ++i) { auto it = labels_map.find(static_cast(decoded_prediction[i])); if (it != labels_map.end()) hypotheses.push_back(it->second); } return hypotheses; } }; static string predict(Mat& features, dnn::Net net, Decoder decoder) { // Passes the features through the Jasper model and decodes the output to english transcripts. // expand 2d features matrix to 3d vector sizes = {1, static_cast(features.size[0]), static_cast(features.size[1])}; features = features.reshape(0, sizes); // make prediction net.setInput(features); Mat output = net.forward(); // decode output to transcript auto prediction = decoder.decode(output); return prediction; } static int readAudioFile(vector& inputAudio, string file, int audioStream) { VideoCapture cap; int samplingRate = 16000; vector params { CAP_PROP_AUDIO_STREAM, audioStream, CAP_PROP_VIDEO_STREAM, -1, CAP_PROP_AUDIO_DATA_DEPTH, CV_32F, CAP_PROP_AUDIO_SAMPLES_PER_SECOND, samplingRate }; cap.open(file, CAP_ANY, params); if (!cap.isOpened()) { cerr << "Error : Can't read audio file: '" << file << "' with audioStream = " << audioStream << endl; return -1; } const int audioBaseIndex = (int)cap.get(CAP_PROP_AUDIO_BASE_INDEX); vector frameVec; Mat frame; for (;;) { if (cap.grab()) { cap.retrieve(frame, audioBaseIndex); frameVec = frame; inputAudio.insert(inputAudio.end(), frameVec.begin(), frameVec.end()); } else { break; } } return samplingRate; } static int readAudioMicrophone(vector& inputAudio, int microTime) { VideoCapture cap; int samplingRate = 16000; vector params { CAP_PROP_AUDIO_STREAM, 0, CAP_PROP_VIDEO_STREAM, -1, CAP_PROP_AUDIO_DATA_DEPTH, CV_32F, CAP_PROP_AUDIO_SAMPLES_PER_SECOND, samplingRate }; cap.open(0, CAP_ANY, params); if (!cap.isOpened()) { cerr << "Error: Can't open microphone" << endl; return -1; } const int audioBaseIndex = (int)cap.get(CAP_PROP_AUDIO_BASE_INDEX); vector frameVec; Mat frame; if (microTime <= 0) { cerr << "Error: Duration of audio chunk must be > 0" << endl; return -1; } size_t sizeOfData = static_cast(microTime * samplingRate); while (inputAudio.size() < sizeOfData) { if (cap.grab()) { cap.retrieve(frame, audioBaseIndex); frameVec = frame; inputAudio.insert(inputAudio.end(), frameVec.begin(), frameVec.end()); } else { cerr << "Error: Grab error" << endl; break; } } return samplingRate; } int main(int argc, char** argv) { const String keys = "{help h usage ? | | This script runs Jasper Speech recognition model }" "{input_file i | | Path to input audio file. If not specified, microphone input will be used }" "{audio_duration t | 15 | Duration of audio chunk to be captured from microphone }" "{audio_stream a | 0 | CAP_PROP_AUDIO_STREAM value }" "{show_spectrogram s | false | Show a spectrogram of the input audio: true / false / 1 / 0 }" "{model m | jasper.onnx | Path to the onnx file of Jasper. You can download the converted onnx model " "from https://drive.google.com/drive/folders/1wLtxyao4ItAg8tt4Sb63zt6qXzhcQoR6?usp=sharing}" "{backend b | dnn::DNN_BACKEND_DEFAULT | Select a computation backend: " "dnn::DNN_BACKEND_DEFAULT, " "dnn::DNN_BACKEND_INFERENCE_ENGINE, " "dnn::DNN_BACKEND_OPENCV }" "{target t | dnn::DNN_TARGET_CPU | Select a target device: " "dnn::DNN_TARGET_CPU, " "dnn::DNN_TARGET_OPENCL, " "dnn::DNN_TARGET_OPENCL_FP16 }" ; CommandLineParser parser(argc, argv, keys); if (parser.has("help")) { parser.printMessage(); return 0; } // Load Network dnn::Net net = dnn::readNetFromONNX(parser.get("model")); net.setPreferableBackend(parser.get("backend")); net.setPreferableTarget(parser.get("target")); // Get audio vectorinputAudio = {}; int samplingRate = 0; if (parser.has("input_file")) { string audio = samples::findFile(parser.get("input_file")); samplingRate = readAudioFile(inputAudio, audio, parser.get("audio_stream")); } else { samplingRate = readAudioMicrophone(inputAudio, parser.get("audio_duration")); } if ((inputAudio.size() == 0) || samplingRate <= 0) { cerr << "Error: problems with audio reading, check input arguments" << endl; return -1; } if (inputAudio.size() / samplingRate < 6) { cout << "Warning: For predictable network performance duration of audio must exceed 6 sec." " Audio will be extended with zero samples" << endl; for(int i = static_cast(inputAudio.size()) - 1; i < samplingRate * 6; ++i) { inputAudio.push_back(0); } } // Calculate features FilterbankFeatures filter; auto calculated_features = filter.calculate_features(inputAudio); // Show spectogram if required if (parser.get("show_spectrogram") == true) { Mat spectogram; normalize(calculated_features, spectogram, 0, 255, NORM_MINMAX, CV_8U); applyColorMap(spectogram, spectogram, COLORMAP_INFERNO); imshow("spectogram", spectogram); waitKey(0); } Decoder decoder; string prediction = predict(calculated_features, net, decoder); for( auto &transcript: prediction) { cout << transcript; } return 0; }