import numpy as np import cv2 as cv import argparse import os import soundfile as sf # Temporary import to load audio files ''' You can download the converted onnx model from https://drive.google.com/drive/folders/1wLtxyao4ItAg8tt4Sb63zt6qXzhcQoR6?usp=sharing or convert the model yourself. You can get the original pre-trained Jasper model from NVIDIA : https://ngc.nvidia.com/catalog/models/nvidia:jasper_pyt_onnx_fp16_amp/files Download and unzip : `$ wget --content-disposition https://api.ngc.nvidia.com/v2/models/nvidia/jasper_pyt_onnx_fp16_amp/versions/20.10.0/zip -O jasper_pyt_onnx_fp16_amp_20.10.0.zip && unzip -o ./jasper_pyt_onnx_fp16_amp_20.10.0.zip && unzip -o ./jasper_pyt_onnx_fp16_amp.zip` you can get the script to convert the model here : https://gist.github.com/spazewalker/507f1529e19aea7e8417f6e935851a01 You can convert the model using the following steps: 1. Import onnx and load the original model ``` import onnx model = onnx.load("./jasper-onnx/1/model.onnx") ``` 3. Change data type of input layer ``` inp = model.graph.input[0] model.graph.input.remove(inp) inp.type.tensor_type.elem_type = 1 model.graph.input.insert(0,inp) ``` 4. Change the data type of output layer ``` out = model.graph.output[0] model.graph.output.remove(out) out.type.tensor_type.elem_type = 1 model.graph.output.insert(0,out) ``` 5. Change the data type of every initializer and cast it's values from FP16 to FP32 ``` for i,init in enumerate(model.graph.initializer): model.graph.initializer.remove(init) init.data_type = 1 init.raw_data = np.frombuffer(init.raw_data, count=np.product(init.dims), dtype=np.float16).astype(np.float32).tobytes() model.graph.initializer.insert(i,init) ``` 6. Add an additional reshape node to handle the inconsistant input from python and c++ of openCV. see https://github.com/opencv/opencv/issues/19091 Make & insert a new node with 'Reshape' operation & required initializer ``` tensor = numpy_helper.from_array(np.array([0,64,-1]),name='shape_reshape') model.graph.initializer.insert(0,tensor) node = onnx.helper.make_node(op_type='Reshape',inputs=['input__0','shape_reshape'], outputs=['input_reshaped'], name='reshape__0') model.graph.node.insert(0,node) model.graph.node[1].input[0] = 'input_reshaped' ``` 7. Finally save the model ``` with open('jasper_dynamic_input_float.onnx','wb') as f: onnx.save_model(model,f) ``` Original Repo : https://github.com/NVIDIA/DeepLearningExamples/tree/master/PyTorch/SpeechRecognition/Jasper ''' class FilterbankFeatures: def __init__(self, sample_rate=16000, window_size=0.02, window_stride=0.01, n_fft=512, preemph=0.97, n_filt=64, lowfreq=0, highfreq=None, log=True, dither=1e-5): ''' 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 ''' self.win_length = int(sample_rate * window_size) # frame size self.hop_length = int(sample_rate * window_stride) # stride self.n_fft = n_fft or 2 ** np.ceil(np.log2(self.win_length)) self.log = log self.dither = dither self.n_filt = n_filt self.preemph = preemph highfreq = highfreq or sample_rate / 2 self.window_tensor = np.hanning(self.win_length) self.filterbanks = self.mel(sample_rate, self.n_fft, n_mels=n_filt, fmin=lowfreq, fmax=highfreq) self.filterbanks.dtype=np.float32 self.filterbanks = np.expand_dims(self.filterbanks,0) def normalize_batch(self, x, seq_len): ''' Normalizes the features. ''' x_mean = np.zeros((seq_len.shape[0], x.shape[1]), dtype=x.dtype) x_std = np.zeros((seq_len.shape[0], x.shape[1]), dtype=x.dtype) for i in range(x.shape[0]): x_mean[i, :] = np.mean(x[i, :, :seq_len[i]],axis=1) x_std[i, :] = np.std(x[i, :, :seq_len[i]],axis=1) # make sure x_std is not zero x_std += 1e-10 return (x - np.expand_dims(x_mean,2)) / np.expand_dims(x_std,2) def calculate_features(self, x, seq_len): ''' Calculates filterbank features. args: x : mono channel audio seq_len : length of the audio sample returns: x : filterbank features ''' dtype = x.dtype seq_len = np.ceil(seq_len / self.hop_length) seq_len = np.array(seq_len,dtype=np.int32) # dither if self.dither > 0: x += self.dither * np.random.randn(*x.shape) # do preemphasis if self.preemph is not None: x = np.concatenate( (np.expand_dims(x[0],-1), x[1:] - self.preemph * x[:-1]), axis=0) # Short Time Fourier Transform x = self.stft(x, n_fft=self.n_fft, hop_length=self.hop_length, win_length=self.win_length, fft_window=self.window_tensor) # get power spectrum x = (x**2).sum(-1) # dot with filterbank energies x = np.matmul(np.array(self.filterbanks,dtype=x.dtype), x) # log features if required if self.log: x = np.log(x + 1e-20) # normalize if required x = self.normalize_batch(x, seq_len).astype(dtype) return x # Mel Frequency calculation def hz_to_mel(self, frequencies): ''' Converts frequencies from hz to mel scale. Input can be a number or a vector. ''' frequencies = np.asanyarray(frequencies) f_min = 0.0 f_sp = 200.0 / 3 mels = (frequencies - f_min) / f_sp # Fill in the log-scale part min_log_hz = 1000.0 # beginning of log region (Hz) min_log_mel = (min_log_hz - f_min) / f_sp # same (Mels) logstep = np.log(6.4) / 27.0 # step size for log region if frequencies.ndim: # If we have array data, vectorize log_t = frequencies >= min_log_hz mels[log_t] = min_log_mel + np.log(frequencies[log_t] / min_log_hz) / logstep elif frequencies >= min_log_hz: # If we have scalar data, directly mels = min_log_mel + np.log(frequencies / min_log_hz) / logstep return mels def mel_to_hz(self, mels): ''' Converts frequencies from mel to hz scale. Input can be a number or a vector. ''' mels = np.asanyarray(mels) # Fill in the linear scale f_min = 0.0 f_sp = 200.0 / 3 freqs = f_min + f_sp * mels # And now the nonlinear scale min_log_hz = 1000.0 # beginning of log region (Hz) min_log_mel = (min_log_hz - f_min) / f_sp # same (Mels) logstep = np.log(6.4) / 27.0 # step size for log region if mels.ndim: # If we have vector data, vectorize log_t = mels >= min_log_mel freqs[log_t] = min_log_hz * np.exp(logstep * (mels[log_t] - min_log_mel)) elif mels >= min_log_mel: # If we have scalar data, check directly freqs = min_log_hz * np.exp(logstep * (mels - min_log_mel)) return freqs def mel_frequencies(self, n_mels=128, fmin=0.0, fmax=11025.0): ''' Calculates n mel frequencies between 2 frequencies args: n_mels : number of bands fmin : min frequency fmax : max frequency returns: mels : vector of mel frequencies ''' # 'Center freqs' of mel bands - uniformly spaced between limits min_mel = self.hz_to_mel(fmin) max_mel = self.hz_to_mel(fmax) mels = np.linspace(min_mel, max_mel, n_mels) return self.mel_to_hz(mels) def mel(self, sr, n_fft, n_mels=128, fmin=0.0, fmax=None, dtype=np.float32): ''' Generates mel filterbank args: sr : Sampling rate n_fft : number of FFT components n_mels : number of Mel bands to generate fmin : lowest frequency (in Hz) fmax : highest frequency (in Hz). sr/2.0 if None dtype : the data type of the output basis. returns: mels : Mel transform matrix ''' # default Max freq = half of sampling rate if fmax is None: fmax = float(sr) / 2 # Initialize the weights n_mels = int(n_mels) weights = np.zeros((n_mels, int(1 + n_fft // 2)), dtype=dtype) # Center freqs of each FFT bin fftfreqs = np.linspace(0, float(sr) / 2, int(1 + n_fft // 2), endpoint=True) # 'Center freqs' of mel bands - uniformly spaced between limits mel_f = self.mel_frequencies(n_mels + 2, fmin=fmin, fmax=fmax) fdiff = np.diff(mel_f) ramps = np.subtract.outer(mel_f, fftfreqs) for i in range(n_mels): # lower and upper slopes for all bins lower = -ramps[i] / fdiff[i] upper = ramps[i + 2] / fdiff[i + 1] # .. then intersect them with each other and zero weights[i] = np.maximum(0, np.minimum(lower, upper)) # Using Slaney-style mel which is scaled to be approx constant energy per channel enorm = 2.0 / (mel_f[2 : n_mels + 2] - mel_f[:n_mels]) weights *= enorm[:, np.newaxis] return weights # STFT preperation def pad_window_center(self, data, size, axis=-1, **kwargs): ''' Centers the data and pads. args: data : Vector to be padded and centered size : Length to pad data axis : Axis along which to pad and center the data kwargs : arguments passed to np.pad return : centered and padded data ''' kwargs.setdefault("mode", "constant") n = data.shape[axis] lpad = int((size - n) // 2) lengths = [(0, 0)] * data.ndim lengths[axis] = (lpad, int(size - n - lpad)) if lpad < 0: raise Exception( ("Target size ({:d}) must be at least input size ({:d})").format(size, n) ) return np.pad(data, lengths, **kwargs) def frame(self, x, frame_length, hop_length): ''' Slices a data array into (overlapping) frames. args: x : array to frame frame_length : length of frame hop_length : Number of steps to advance between frames return : A framed view of `x` ''' if x.shape[-1] < frame_length: raise Exception( "Input is too short (n={:d})" " for frame_length={:d}".format(x.shape[-1], frame_length) ) x = np.asfortranarray(x) n_frames = 1 + (x.shape[-1] - frame_length) // hop_length strides = np.asarray(x.strides) new_stride = np.prod(strides[strides > 0] // x.itemsize) * x.itemsize shape = list(x.shape)[:-1] + [frame_length, n_frames] strides = list(strides) + [hop_length * new_stride] return np.lib.stride_tricks.as_strided(x, shape=shape, strides=strides) def dtype_r2c(self, d, default=np.complex64): ''' Find the complex numpy dtype corresponding to a real dtype. args: d : The real-valued dtype to convert to complex. default : The default complex target type, if `d` does not match a known dtype return : The complex dtype ''' mapping = { np.dtype(np.float32): np.complex64, np.dtype(np.float64): np.complex128, } dt = np.dtype(d) if dt.kind == "c": return dt return np.dtype(mapping.get(dt, default)) def stft(self, y, n_fft, hop_length=None, win_length=None, fft_window=None, pad_mode='reflect', return_complex=False): ''' Short Time Fourier Transform. The STFT represents a signal in the time-frequency domain by computing discrete Fourier transforms (DFT) over short overlapping windows. args: y : input signal n_fft : length of the windowed signal after padding with zeros. hop_length : number of audio samples between adjacent STFT columns. win_length : Each frame of audio is windowed by window of length win_length and then padded with zeros to match n_fft fft_window : a vector or array of length `n_fft` having values computed by a window function pad_mode : mode while padding the singnal return_complex : returns array with complex data type if `True` return : Matrix of short-term Fourier transform coefficients. ''' if win_length is None: win_length = n_fft if hop_length is None: hop_length = int(win_length // 4) if y.ndim!=1: raise Exception(f'Invalid input shape. Only Mono Channeled audio supported. Input must have shape (Audio,). Got {y.shape}') # Pad the window out to n_fft size fft_window = self.pad_window_center(fft_window, n_fft) # Reshape so that the window can be broadcast fft_window = fft_window.reshape((-1, 1)) # Pad the time series so that frames are centered y = np.pad(y, int(n_fft // 2), mode=pad_mode) # Window the time series. y_frames = self.frame(y, frame_length=n_fft, hop_length=hop_length) # Convert data type to complex dtype = self.dtype_r2c(y.dtype) # Pre-allocate the STFT matrix stft_matrix = np.empty( (int(1 + n_fft // 2), y_frames.shape[-1]), dtype=dtype, order="F") stft_matrix = np.fft.rfft( fft_window * y_frames, axis=0) return stft_matrix if return_complex==True else np.stack((stft_matrix.real,stft_matrix.imag),axis=-1) class Decoder: ''' Used for decoding the output of jasper model. ''' def __init__(self): 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',"'"] self.labels_map = {i: label for i,label in enumerate(labels)} self.blank_id = 28 def decode(self,x): """ Takes output of Jasper model and performs ctc decoding algorithm to remove duplicates and special symbol. Returns prediction """ x = np.argmax(x,axis=-1) hypotheses = [] prediction = x.tolist() # CTC decoding procedure decoded_prediction = [] previous = self.blank_id for p in prediction: if (p != previous or previous == self.blank_id) and p != self.blank_id: decoded_prediction.append(p) previous = p hypothesis = ''.join([self.labels_map[c] for c in decoded_prediction]) hypotheses.append(hypothesis) return hypotheses def predict(features, net, decoder): ''' Passes the features through the Jasper model and decodes the output to english transcripts. args: features : input features, calculated using FilterbankFeatures class net : Jasper model dnn.net object decoder : Decoder object return : Predicted text ''' # This is a workaround https://github.com/opencv/opencv/issues/19091 # expanding 1 dimentions allows us to pass it to the network # from python. This should be resolved in the future. features = np.expand_dims(features,axis=3) # make prediction net.setInput(features) output = net.forward() # decode output to transcript prediction = decoder.decode(output.squeeze(0)) return prediction[0] if __name__ == '__main__': # Computation backends supported by layers backends = (cv.dnn.DNN_BACKEND_DEFAULT, cv.dnn.DNN_BACKEND_INFERENCE_ENGINE, cv.dnn.DNN_BACKEND_OPENCV) # Target Devices for computation targets = (cv.dnn.DNN_TARGET_CPU, cv.dnn.DNN_TARGET_OPENCL, cv.dnn.DNN_TARGET_OPENCL_FP16) parser = argparse.ArgumentParser(description='This script runs Jasper Speech recognition model', formatter_class=argparse.ArgumentDefaultsHelpFormatter) parser.add_argument('--input_audio', type=str, required=True, help='Path to input audio file. OR Path to a txt file with relative path to multiple audio files in different lines') parser.add_argument('--show_spectrogram', action='store_true', help='Whether to show a spectrogram of the input audio.') parser.add_argument('--model', type=str, default='jasper.onnx', help='Path to the onnx file of Jasper. default="jasper.onnx"') parser.add_argument('--output', type=str, help='Path to file where recognized audio transcript must be saved. Leave this to print on console.') parser.add_argument('--backend', choices=backends, default=cv.dnn.DNN_BACKEND_DEFAULT, type=int, help='Select a computation backend: ' "%d: automatically (by default) " "%d: OpenVINO Inference Engine " "%d: OpenCV Implementation " % backends) parser.add_argument('--target', choices=targets, default=cv.dnn.DNN_TARGET_CPU, type=int, help='Select a target device: ' "%d: CPU target (by default) " "%d: OpenCL " "%d: OpenCL FP16 " % targets) args, _ = parser.parse_known_args() if args.input_audio and not os.path.isfile(args.input_audio): raise OSError("Input audio file does not exist") if not os.path.isfile(args.model): raise OSError("Jasper model file does not exist") if args.input_audio.endswith('.txt'): with open(args.input_audio) as f: content = f.readlines() content = [x.strip() for x in content] audio_file_paths = content for audio_file_path in audio_file_paths: if not os.path.isfile(audio_file_path): raise OSError("Audio file({audio_file_path}) does not exist") else: audio_file_paths = [args.input_audio] audio_file_paths = [os.path.abspath(x) for x in audio_file_paths] # Read audio Files features = [] try: for audio_file_path in audio_file_paths: audio = sf.read(audio_file_path) # If audio is stereo, just take one channel. X = audio[0] if audio[0].ndim==1 else audio[0][:,0] features.append(X) except: raise Exception(f"Soundfile cannot read {args.input_audio}. Try a different format") # Get Filterbank Features feature_extractor = FilterbankFeatures() for i in range(len(features)): X = features[i] seq_len = np.array([X.shape[0]], dtype=np.int32) features[i] = feature_extractor.calculate_features(x=X, seq_len=seq_len) # Load Network net = cv.dnn.readNetFromONNX(args.model) net.setPreferableBackend(args.backend) net.setPreferableTarget(args.target) # Show spectogram if required if args.show_spectrogram and not args.input_audio.endswith('.txt'): img = cv.normalize(src=features[0][0], dst=None, alpha=0, beta=255, norm_type=cv.NORM_MINMAX, dtype=cv.CV_8U) img = cv.applyColorMap(img, cv.COLORMAP_JET) cv.imshow('spectogram', img) cv.waitKey(0) # Initialize decoder decoder = Decoder() # Make prediction prediction = [] print("Predicting...") for feature in features: print(f"\rAudio file {len(prediction)+1}/{len(features)}", end='') prediction.append(predict(feature, net, decoder)) print("") # save transcript if required if args.output: with open(args.output,'w') as f: for pred in prediction: f.write(pred+'\n') print("Transcript was written to {}".format(args.output)) else: print(prediction) cv.destroyAllWindows()