# Towards efficient models for real-time deep noise suppression

Contents

## Notes

### ABSTRACT

architectures on large dataset.

• Reverberation was also considered
• Evaluated Convolutional Recurrent Network architectures
• Trade off between computational complexity and the achievable speech quality

### INTRODUCTION

#### Goal

which network parts can be scaled, removed, or replaced by more efficient modules, at which gains in complexity and which loss in quality. Specifically, we investigate the influence of RNN size, type, and the use of disconnected parallel RNNs. For CRNs with a symmetric convolutional encoder/decoder structure, we investigate the convolution layers, spectral vs. spectro-temporal convolutions, and skip connections. As a result, we propose an efficient CRN structure with around 4-5 times less computational operations with similar quality than previously proposed CRNs.

### ENHANCEMENT SYSTEM AND TRAINING OBJECTIVE

• Spectral Suppression based enhancement systems - robust generalization, logical interpretation and control, and easier integration with existing speech processing algorithms.

• Input Features are log power spectra

• Output is a real-valued, time varying suppression gain per time frequency bin

• To compute a single frame, the network requires only the features of the current frame, or when using causal convolution, also several past frames. So latency depends only on the STFT window size.

• STFT consistency is enforced by propagating FD output through reconstruction and another STFT to compute a FD loss.

• Both Predicted and target signals are normalized by the active target utterance level, to enure balanced optimization for signal level dependent losses.

• MSE is used, blending the magnitude only with a phase aware term, which is found to be superior to other losses for reverberant speech enhancement. Loss function is given by $L = (1-\lambda)\Sigma_k,n||S|^c - |\hat S|^c|^2 + \lambda \Sigma_k,n||S|^c e^j\psi s - |\hat S|^c e^j \psi s|^2$ where c = 0.3 is a compression factor, λ = 0.3 is a weighting between complex and magnitude based loss, and omitted the dependency of the target speech spectral bins S(k,n) on the frequency and time indices k, n for brevity.

• The networks are trained in batches of 10 sequences of 10 s length using the AdamW optimizer, learning rate of $8 * 10^-5$ and weight decay of 0.1.

• The best model is picked based on the validation metric using a heuristic optimization criterion using perceptual evaluation of speech quality, scale invariant signal to noise ration and cepstral distance $Q = PESQ + 0.2 * siSRD - CD$

### NETWORK ARCHITECTURES

#### NSnet2

• Consists only of fully connected and gated recurrent unit layers in

the form FC-GRU-GRU-FC-FC-FC.

• All FC layers are followed by ReLU activation
• Except the last layer has sigmoid activations to predict constrained suppression
• The standard layer dimensions are 400 for GRUs and 600 for FC layers i/e/ 400-400-400-600-600-K
• Also investigated different configurations
• The input and output dimensions are the number of frequency bins.

#### CRUSE

• CRN U-Net structure
• Convolutional Recurrent U-net for Speech Enhancement (CRUSE)
• L symmetric convolutional and deconvolutional encoder and decoder layers with kernels of size (2,3) in time an frequency dimensions. Convolution kernels move with a stride of (1,2) i.e. downsample the features along the frequency axis efficiently while the number of channels C_l for layer 1,…,L increase per encoder layer, and decrease mirrored in the decoder. In this work the input and output channels C_in = C_out = 1 but they can be extended to e.g. take complex values or multiple features as multiple channels. Convolutional layers are followed by leaky ReLU activations while the last layer uses sigmoid. Between encoder and decoder sits a recurrent layer, which is fed with all features flattened along channels.
• Using 1 GRU layer over 2 LSTM layers at this stage leads to little performance loss, but huge computational savings
• GRU saves 25% complexity compared to an LSTM layer.

Further Modifications

• Parallel RNN grouping Performance of both CRUSE and NSNet2 directly scales with bottleneck size i.e. the width R of the central RNN layers. However, the complexity of RNN layers scales with R^2 making wide RNNs computational unattractive. There for RNNs are grouped into P disconnected parallel RNNs, still yielding the same forward information flow.

• Possible parallel execution of disconnected RNNs.
• Skip Connections Use skip connection by adding rather than concatenating.

• More effective

Also added a trainable channel wise scaling and bias in the add skip connection implemented as convolutions

### EXPERIMENTAL SETUP

#### Dataset

• large scale synthetic training set and test on real recordings
• 544 hours of high mean opinion score
• Most data open sourced
• T_60] > 0.22s and C_50 < 18 dB implies reverberation
• Data Generation Pipeline

#### Algorithmic parameters

• 16kHz sampling frequency
• STFT with 50% overlapping squareroot-Hann windows of 20ms length
• FFT size of 320 points
• Input is 161 dimensional log power spectra
• NSnet2-R where R denotes the number of GRU node per layer
• parameterize CRUSE with different encoder decoder sizes, starting alway with C_1 = 16 channels, and doubling the channels each layer.
• CRUSE L-C_L - NxRNNP, where L are the number of E-D layers, the last encoder layer filters C_L can vary to scale the RNN layer width, N are the number of RNN layers and P are the number of parallel RNNs.
• Convolution kernels are always (2,3), unless denoted explicitly as 1D convolutions with (1,3) kernels operating only across frequency.

### CONCLUSIONS

• Speech Quality is a function of network size, especially depending on the recurrent layer width

## Bibliography

Braun, Sebastian, Hannes Gamper, Chandan K. A. Reddy, and Ivan Tashev. 2021. “Towards Efficient Models for Real-Time Deep Noise Suppression.” CoRR. http://arxiv.org/abs/2101.09249v2.

Luo, Yi, and Nima Mesgarani. 2019. “Conv-Tasnet: Surpassing Ideal Time-Frequency Magnitude Masking for Speech Separation.” IEEE/ACM Transactions on Audio, Speech, and Language Processing 27 (8):1256–66. https://doi.org/10.1109/taslp.2019.2915167.