Lessons from Building Acoustic Models with a Million Hours of Speech

This is a report of our lessons learned building acoustic models from 1 Million hours of unlabeled speech, while labeled speech is restricted to 7,000 hours. We employ student/teacher training on unlabeled data, helping scale out target generation in comparison to confidence model based methods, which require a decoder and a confidence model. To optimize storage and to parallelize target generation, we store high valued logits from the teacher model. Introducing the notion of scheduled learning, we interleave learning on unlabeled and labeled data. To scale distributed training across a large number of GPUs, we use BMUF with 64 GPUs, while performing sequence training only on labeled data with gradient threshold compression SGD using 16 GPUs. Our experiments show that extremely large amounts of data are indeed useful; with little hyper-parameter tuning, we obtain relative WER improvements in the 10 to 20% range, with higher gains in noisier conditions.Comment: "Copyright 2019 IEEE. Personal use of this material is permitted. Permission from IEEE must be obtained for all other uses, in any current or future media, including reprinting/republishing this material for advertising or promotional purposes, creating new collective works, for resale or redistribution to servers or lists, or reuse of any copyrighted component of this work in other works.

Privacy-Sensitive Audio Features for Conversational Speech Processing

The work described in this thesis takes place in the context of capturing real-life audio for the analysis of spontaneous social interactions. Towards this goal, we wish to capture conversational and ambient sounds using portable audio recorders. Analysis of conversations can then proceed by modeling the speaker turns and durations produced by speaker diarization. However, a key factor against the ubiquitous capture of real-life audio is privacy. Particularly, recording and storing raw audio would breach the privacy of people whose consent has not been explicitly obtained. In this thesis, we study audio features instead – for recording and storage – that can respect privacy by minimizing the amount of linguistic information, while achieving state-of-the-art performance in conversational speech processing tasks. Indeed, the main contributions of this thesis are the achievement of state-of-the-art performances in speech/nonspeech detection and speaker diarization tasks using such features, which we refer to, as privacy-sensitive. Besides this, we provide a comprehensive analysis of these features for the two tasks in a variety of conditions, such as indoor (predominantly) and outdoor audio. To objectively evaluate the notion of privacy, we propose the use of human and automatic speech recognition tests, with higher accuracy in either being interpreted as yielding lower privacy. For the speech/nonspeech detection (SND) task, this thesis investigates three different approaches to privacy-sensitive features. These approaches are based on simple, instantaneous, feature extraction methods, excitation source information based methods, and feature obfuscation methods. These approaches are benchmarked against Perceptual Linear Prediction (PLP) features under many conditions on a large meeting dataset of nearly 450 hours. Additionally, automatic speech (phoneme) recognition studies on TIMIT showed that the proposed features yield low phoneme recognition accuracies, implying higher privacy. For the speaker diarization task, we interpret the extraction of privacy-sensitive features as an objective that maximizes the mutual information (MI) with speakers while minimizing the MI with phonemes. The source-filter model arises naturally out of this formulation. We then investigate two different approaches for extracting excitation source based features, namely Linear Prediction (LP) residual and deep neural networks. Diarization experiments on the single and multiple distant microphone scenarios from the NIST rich text evaluation datasets show that these features yield a performance close to the Mel Frequency Cepstral coefficients (MFCC) features. Furthermore, listening tests support the proposed approaches in terms of yielding low intelligibility in comparison with MFCC features. The last part of the thesis studies the application of our methods to SND and diarization in outdoor settings. While our diarization study was more preliminary in nature, our study on SND brings about the conclusion that privacy-sensitive features trained on outdoor audio yield performance comparable to that of PLP features trained on outdoor audio. Lastly, we explored the suitability of using SND models trained on indoor conditions for the outdoor audio. Such an acoustic mismatch caused a large drop in performance, which could not be compensated even by combining indoor models

Exploiting contextual information for speech/non-speech detection

In this paper, we investigate the effect of temporal context for speech/non-speech detection (SND). It is shown that even a simple feature such as full-band energy, when employed with a large-enough context, shows promise for further investigation. Experimental evaluations on the test data set, with a state-of-the-art multi-layer perceptron based SND system and a simple energy threshold based SND method, using the F-measure, show an absolute performance gain of $4.4\%$ and $5.4\%$ respectively. The optimal contextual length was found to be 1000 ms. Further numerical optimizations yield an improvement ($3.37\%$ absolute), resulting in an absolute gain of $7.77\%$ and $8.77\%$ over the MLP based and energy based methods respectively. ROC based performance evaluation also reveals promising performance for the proposed method, particularly in low SNR conditions

Exploiting Contextual Information for Speech/Non-Speech Detection

In this paper, we investigate the effect of temporal context for speech/non-speech detection (SND). It is shown that even a simple feature such as full-band energy, when employed with a large-enough context, shows promise for further investigation. Experimental evaluations on the test data set, with a state-of-the-art multi-layer perceptron based SND system and a simple energy threshold based SND method, using the F-measure, show an absolute performance gain of 4.4% and 5.4% respectively. The optimal contextual length was found to be 1000 ms. Further numerical optimizations yield an improvement (3.37% absolute), resulting in an absolute gain of 7.77% and 8.77% over the MLP based and energy based methods respectively. ROC based performance evaluation also reveals promising performance for the proposed method, particularly in low SNR conditions

Exploiting temporal context for speech/non-speech detection

In this paper, we investigate the effect of temporal context for speech/non-speech detection (SND). It is shown that even a simple feature such as full-band energy, when employed with a large-enough context, shows promise for further investigation. Experimental evaluations on the test data set, with a state-of-the-art multi-layer perceptron based SND system and a simple energy threshold based SND method, using the F-measure, show an absolute performance gain of $4.4\%$ and $5.4\%$ respectively, when used with a context of 1000 ms. ROC based performance evaluation also reveals promising performance for the proposed method, particularly in low SNR conditions

Wordless Sounds: Robust Speaker Diarization using Privacy-Preserving Audio Representations

This paper investigates robust privacy-sensitive audio features for speaker diarization in multiparty conversations: ie., a set of audio features having low linguistic information for speaker diarization in a single and multiple distant microphone scenarios. We systematically investigate Linear Prediction (LP) residual. Issues such as prediction order and choice of representation of LP residual are studied. Additionally, we explore the combination of LP residual with subband information from 2.5 kHz to 3.5 kHz and spectral slope. Next, we propose a supervised framework using deep neural architecture for deriving privacy-sensitive audio features. We benchmark these approaches against the traditional Mel Frequency Cepstral Coefficients (MFCC) features for speaker diarization in both the microphone scenarios. Experiments on the RT07 evaluation dataset show that the proposed approaches yield diarization performance close to the MFCC features on the single distant microphone dataset. To objectively evaluate the notion of privacy in terms of linguistic information, we perform human and automatic speech recognition tests, showing that the proposed approaches to privacy-sensitive audio features yield much lower recognition accuracies compared to MFCC features

LP Residual Features for Robust, Privacy-Sensitive Speaker Diarization

We present a comprehensive study of linear prediction residual for speaker diarization on single and multiple distant microphone conditions in privacy-sensitive settings, a requirement to analyze a wide range of spontaneous conversations. Two representations of the residual are compared, namely real-cepstrum and MFCC, with the latter performing better. Experiments on RT06eval show that residual with subband information from 2.5 kHz to 3.5 kHz and spectral slope yields a performance close to traditional MFCC features. As a way to objectively evaluate privacy in terms of linguistic information, we perform phoneme recognition. Residual features yield low phoneme accuracies compared to traditional MFCC features

Robustness of Phase based Features for Speaker Recognition

This paper demonstrates the robustness of group-delay based features for speech processing. An analysis of group delay functions is presented which show that these features retain formant structure even in noise. Furthermore, a speaker verification task performed on the NIST 2003 database show lesser error rates, when compared with the traditional MFCC features. We also mention about using feature diversity to dynamically choose the feature for every claimed speaker

Robustness of Group Delay Representations for Noisy Speech Signals

This paper demonstrates the robustness of group delay based features to additive noise. First, we analytically show the robustness of group delay based represen- tations. The analysis makes use of the fact that, for minimum-phase signals, the group delay function can be represented in terms of the cepstral coefficients of the log-magnitude spectrum. Such a representation results in the speech spectrum dominating over the noise spectrum, both at low and high SNRs. Further, we ex- perimentally demonstrate the robustness of the representation on a voice activity detection (VAD) task, comparing a group delay based VAD algorithm with standard VAD methods as well as a magnitude-spectrum based method