Deep Learning for Audio Signal Processing

Given the recent surge in developments of deep learning, this article provides a review of the state-of-the-art deep learning techniques for audio signal processing. Speech, music, and environmental sound processing are considered side-by-side, in order to point out similarities and differences between the domains, highlighting general methods, problems, key references, and potential for cross-fertilization between areas. The dominant feature representations (in particular, log-mel spectra and raw waveform) and deep learning models are reviewed, including convolutional neural networks, variants of the long short-term memory architecture, as well as more audio-specific neural network models. Subsequently, prominent deep learning application areas are covered, i.e. audio recognition (automatic speech recognition, music information retrieval, environmental sound detection, localization and tracking) and synthesis and transformation (source separation, audio enhancement, generative models for speech, sound, and music synthesis). Finally, key issues and future questions regarding deep learning applied to audio signal processing are identified.Comment: 15 pages, 2 pdf figure

Music Structure Boundaries Estimation Using Multiple Self-Similarity Matrices as Input Depth of Convolutional Neural Networks

International audienceIn this paper, we propose a new representation as input of a Convolutional Neural Network with the goal of estimating music structure boundaries. For this task, previous works used a network performing the late-fusion of a Mel-scaled log-magnitude spectrogram and a self-similarity-lag-matrix. We propose here to use the square-sub-matrices centered on the main diagonals of several self-similarity-matrices, each one representing a different audio descriptors. We propose to combine them using the depth of the input layer. We show that this representation improves the results over the use of the self-similarity-lag-matrix. We also show that using the depth of the input layer provide a convenient way for early fusion of audio representations

16th Sound and Music Computing Conference SMC 2019 (28–31 May 2019, Malaga, Spain)

The 16th Sound and Music Computing Conference (SMC 2019) took place in Malaga, Spain, 28-31 May 2019 and it was organized by the Application of Information and Communication Technologies Research group (ATIC) of the University of Malaga (UMA). The SMC 2019 associated Summer School took place 25-28 May 2019. The First International Day of Women in Inclusive Engineering, Sound and Music Computing Research (WiSMC 2019) took place on 28 May 2019. The SMC 2019 TOPICS OF INTEREST included a wide selection of topics related to acoustics, psychoacoustics, music, technology for music, audio analysis, musicology, sonification, music games, machine learning, serious games, immersive audio, sound synthesis, etc

Deep Learning for Environmentally Robust Speech Recognition: An Overview of Recent Developments

Eliminating the negative effect of non-stationary environmental noise is a long-standing research topic for automatic speech recognition that stills remains an important challenge. Data-driven supervised approaches, including ones based on deep neural networks, have recently emerged as potential alternatives to traditional unsupervised approaches and with sufficient training, can alleviate the shortcomings of the unsupervised methods in various real-life acoustic environments. In this light, we review recently developed, representative deep learning approaches for tackling non-stationary additive and convolutional degradation of speech with the aim of providing guidelines for those involved in the development of environmentally robust speech recognition systems. We separately discuss single- and multi-channel techniques developed for the front-end and back-end of speech recognition systems, as well as joint front-end and back-end training frameworks

Final Research Report on Auto-Tagging of Music

The deliverable D4.7 concerns the work achieved by IRCAM until M36 for the “auto-tagging of music”. The deliverable is a research report. The software libraries resulting from the research have been integrated into Fincons/HearDis! Music Library Manager or are used by TU Berlin. The final software libraries are described in D4.5. The research work on auto-tagging has concentrated on four aspects: 1) Further improving IRCAM’s machine-learning system ircamclass. This has been done by developing the new MASSS audio features, including audio augmentation and audio segmentation into ircamclass. The system has then been applied to train HearDis! “soft” features (Vocals-1, Vocals-2, Pop-Appeal, Intensity, Instrumentation, Timbre, Genre, Style). This is described in Part 3. 2) Developing two sets of “hard” features (i.e. related to musical or musicological concepts) as specified by HearDis! (for integration into Fincons/HearDis! Music Library Manager) and TU Berlin (as input for the prediction model of the GMBI attributes). Such features are either derived from previously estimated higher-level concepts (such as structure, key or succession of chords) or by developing new signal processing algorithm (such as HPSS) or main melody estimation. This is described in Part 4. 3) Developing audio features to characterize the audio quality of a music track. The goal is to describe the quality of the audio independently of its apparent encoding. This is then used to estimate audio degradation or music decade. This is to be used to ensure that playlists contain tracks with similar audio quality. This is described in Part 5. 4) Developing innovative algorithms to extract specific audio features to improve music mixes. So far, innovative techniques (based on various Blind Audio Source Separation algorithms and Convolutional Neural Network) have been developed for singing voice separation, singing voice segmentation, music structure boundaries estimation, and DJ cue-region estimation. This is described in Part 6.EC/H2020/688122/EU/Artist-to-Business-to-Business-to-Consumer Audio Branding System/ABC D

Relative Positional Encoding for Transformers with Linear Complexity

Recent advances in Transformer models allow for unprecedented sequence lengths, due to linear space and time complexity. In the meantime, relative positional encoding (RPE) was proposed as beneficial for classical Transformers and consists in exploiting lags instead of absolute positions for inference. Still, RPE is not available for the recent linear-variants of the Transformer, because it requires the explicit computation of the attention matrix, which is precisely what is avoided by such methods. In this paper, we bridge this gap and present Stochastic Positional Encoding as a way to generate PE that can be used as a replacement to the classical additive (sinusoidal) PE and provably behaves like RPE. The main theoretical contribution is to make a connection between positional encoding and cross-covariance structures of correlated Gaussian processes. We illustrate the performance of our approach on the Long-Range Arena benchmark and on music generation.Comment: ICML 2021 (long talk) camera-ready. 24 page

Group-wise automatic music transcription

Background: Music transcription is the conversion of musical audio into notation such that a musician can recreate the piece. Automatic music transcription (AMT) is the automation of this process. Current AMT algorithms produce a less musically meaningful transcription than human transcribers. However, AMT performs better at predicting notes present in a short time frame. Group-wise Automatic Music Transcription, (GWAMT) is when several renditions of a piece are used to give a single transcription. Aims: The main aim was to perform investigations into GWAMT. Secondary aims included: Comparing methods for GWAMT on the frame level; Considering the impact of GWAMT on the broader field of AMT. Method(s)/Procedures: GWAMT transcription is split into three stages: transcription, alignment and combination. Transcription is performed by splitting the piece into frames, and using a classifier to identify the notes present. Convolutional Neural Networks (CNNs) are used with a novel training methodology and architecture. Different renditions of the same piece have corresponding notes occurring at different times. In order to match corresponding frames, methods for the alignment of multiple renditions are used. Several methods were compared, pairwise alignment, progressive alignment and a new method, iterative alignment. The effect of when the aligned features are combined (early/late), and how (majority vote, linear opinion pool, logarithmic opinion pool, max, median), is investigated. Results: The developed method for frame-level transcription achieves state-of-the-art transcription accuracy on the MAPS database with an F1-score of 76.67%. Experiments on GWAMT show that the F1-score can be improved by between 0.005 to 0.01 using the majority vote and logarithmic pool combination methods. Conclusions/Implications: These experiments show that group-wise frame-level transcription can improve the transcription when there are different tempos, noise levels, dynamic ranges and reverbs between the clips. They also demonstrate a future application of GWAMT to individual pieces with repeated segments