Company Overview

Home / Company Overview

About AppTek

AppTek develops engines and solutions for cross lingual communication. Apptek is a leader in automatic speech recognition (ASR), neural machine translation (MT), machine learning (ML), natural language understanding (NLU) and artificial intelligence (AI).   Founded in 1990, AppTek employs one of the most agile, talented teams of ASR, MT, and NLU PhD scientists and research engineers in the world.  Through our advanced research in speech recognition, machine translation and artificial intelligence, we have solved many challenging problems improving human-quality transcription, language understanding and translation accuracy.  Our people are among the language technology and machine learning industry’s premier experts. Our long-standing affiliations with the world’s leading human language technology universities is central to our continuous introduction of new theories and solutions for automating recognition, translation and communication. Our 30-year history of achieving performance goals with our customers across government, global commerce, call centers and media comes from our understanding of their problems and the best application of technology solutions.

Company History and Timeline

1990
Company founded to enable global multilingual commerce applications and communications
2001
Began providing critical support to USG with advanced text analytics and machine translation software
2002
Patent filed for Automatic Speech Recognition method
2009
USG awards AppTek first multi-speaker, multilingual auto speech and translation system
2015
Patent for Keyword Speech Recognition
2014
eBay acquires AppTek's Hybrid Machine Translation platform for cross-border trade
2015
Launch of new AI platform for ASR and NMT
2016
Two Patents for Deep Neural Network Model Advancements
2018
Patent for Audio Recognition of Keywords
2019
AppTek Wins Two 2019 SpeechTEK People’s Choice Awards
2019
Hermann Ney, Science Director, granted IEEE's James L Flanagan award for pioneering life-long advancements in speech technology

Recent Academic Research and Publications

Equivalence of Segmental and Neural Transducer Modeling: A Proof of Concept

June 2021
Wei Zhou, Albert Zeyer, André Merboldt, Ralf Schlüter, Hermann Ney

With the advent of direct models in automatic speech recognition (ASR), the formerly prevalent frame-wise acoustic modeling based on hidden Markov models (HMM) diversified into a number of modeling architectures like encoder-decoder attention models, transducer models and segmental models (direct HMM). While transducer models stay with a frame-level model definition, segmental models are defined on the level of label segments, directly. While (soft-)attention-based models avoid explicit alignment, transducer and segmental approach internally do model alignment, either by segment hypotheses or, more implicitly, by emitting so-called blank symbols. In this work, we prove that the widely used class of RNN-Transducer models and segmental models (direct HMM) are equivalent and therefore show equal modeling power. It is shown that blank probabilities translate into segment length probabilities and vice versa. In addition, we provide initial experiments investigating decoding and beam-pruning, comparing time-synchronous and label-/segment-synchronous search strategies and their properties using the same underlying model.

View Research

On Sampling-Based Training Criteria for Neural Language Modeling

April 2021
Yingbo Gao, David Thulke, Alexander Gerstenberger, Khoa Viet Tran, Ralf Schlüter, Hermann Ney

As the vocabulary size of modern word-based language models becomes ever larger, many sampling-based training criteria are proposed and investigated. The essence of these sampling methods is that the softmax-related traversal over the entire vocabulary can be simplified, giving speedups compared to the baseline. A problem we notice about the current landscape of such sampling methods is the lack of a systematic comparison and some myths about preferring one over another. In this work, we consider Monte Carlo sampling, importance sampling, a novel method we call compensated partial summation, and noise contrastive estimation. Linking back to the three traditional criteria, namely mean squared error, binary cross-entropy, and cross-entropy, we derive the theoretical solutions to the training problems. Contrary to some common belief, we show that all these sampling methods can perform equally well, as long as we correct for the intended class posterior probabilities. Experimental results in language modeling and automatic speech recognition on Switchboard and LibriSpeech support our claim, with all sampling-based methods showing similar perplexities and word error rates while giving the expected speedups.

View Research

The Impact of ASR on the Automatic Analysis of Linguistic Complexity and Sophistication in Spontaneous L2 Speech

April 2021
Yu Qiao, Zhou Wei, Elma Kerz, Ralf Schlüter

In recent years, automated approaches to assessing linguistic complexity in second language (L2) writing have made significant progress in gauging learner performance, predicting human ratings of the quality of learner productions, and benchmarking L2 development. In contrast, there is comparatively little work in the area of speaking, particularly with respect to fully automated approaches to assessing L2 spontaneous speech. While the importance of a well-performing ASR system is widely recognized, little research has been conducted to investigate the impact of its performance on subsequent automatic text analysis. In this paper, we focus on this issue and examine the impact of using a state-of-the-art ASR system for subsequent automatic analysis of linguistic complexity in spontaneously produced L2 speech. A set of 34 selected measures were considered, falling into four categories: syntactic, lexical, n-gram frequency, and information-theoretic measures. The agreement between the scores for these measures obtained on the basis of ASR-generated vs. manual transcriptions was determined through correlation analysis. A more differential effect of ASR performance on specific types of complexity measures when controlling for task type effects is also presented.

View Research

Comparing the Benefit of Synthetic Training Data for Various Automatic Speech Recognition Architectures

April 2021
Nick Rossenbach, Mohammad Zeineldeen, Benedikt Hilmes, Ralf Schlüter, Hermann Ney

Recent publications on automatic-speech-recognition (ASR) have a strong focus on attention encoder-decoder (AED) architectures which work well for large datasets, but tend to overfit when applied in low resource scenarios. One solution to tackle this issue is to generate synthetic data with a trained text-to-speech system (TTS) if additional text is available. This was successfully applied in many publications with AED systems. We present a novel approach of silence correction in the data pre-processing for TTS systems which increases the robustness when training on corpora targeted for ASR applications. In this work we do not only show the successful application of synthetic data for AED systems, but also test the same method on a highly optimized state-of-the-art Hybrid ASR system and a competitive monophone based system using connectionist-temporal-classification (CTC). We show that for the later systems the addition of synthetic data only has a minor effect, but they still outperform the AED systems by a large margin on LibriSpeech-100h. We achieve a final word-error-rate of 3.3%/10.0% with a Hybrid system on the clean/noisy test-sets, surpassing any previous state-of-the-art systems that do not include unlabeled audio data.

View Research

Acoustic Data-Driven Subword Modeling for End-to-End Speech Recognition

April 2021
Wei Zhou, Mohammad Zeineldeen, Zuoyun Zheng, Ralf Schlüter, Hermann Ney

Subword units are commonly used for end-to-end automatic speech recognition (ASR), while a fully acoustic-oriented subword modeling approach is somewhat missing. We propose an acoustic data-driven subword modeling (ADSM) approach that adapts the advantages of several text-based and acoustic-based subword methods into one pipeline. With a fully acoustic-oriented label design and learning process, ADSM produces acoustic-structured subword units and acoustic-matched target sequence for further ASR training. The obtained ADSM labels are evaluated with different end-to-end ASR approaches including CTC, RNN-transducer and attention models. Experiments on the LibriSpeech corpus show that ADSM clearly outperforms both byte pair encoding (BPE) and pronunciation-assisted subword modeling (PASM) in all cases. Detailed analysis shows that ADSM achieves acoustically more logical word segmentation and more balanced sequence length, and thus, is suitable for both time-synchronous and label-synchronous models. We also briefly describe how to apply acoustic-based subword regularization and unseen text segmentation using ADSM.

View Research

Librispeech Transducer Model with Internal Language Model Prior Correction

April 2021
Albert Zeyer, André Merboldt, Wilfried Michel, Ralf Schlüter, Hermann Ney

is justified by a Bayesian interpretation where the transducer model prior is given by the estimated internal LM. The subtraction of the internal LM gives us over 14% relative improvement over normal shallow fusion. Our transducer has a separate probability distribution for the non-blank labels which allows for easier combination with the external LM, and easier estimation of the internal LM. We additionally take care of including the end-of-sentence (EOS) probability of the external LM in the last blank probability which further improves the performance. All our code and setups are published.

View Research

A New Training Pipeline for an Improved Neural Transducer

May 2020
Albert Zeyer | André Merboldt | Ralf Schlüter | Hermann Ney

The RNN transducer is a promising end-to-end model candidate. We compare the original training criterion with the full marginalization over all alignments, to the commonly used maximum approximation, which simplifies, improves and speeds up our training. We also generalize from the original neural network model and study more powerful models, made possible due to the maximum approximation. We further generalize the output label topology to cover RNN-T, RNA and CTC. We perform several studies among all these aspects, including a study on the effect of external alignments. We find that the transducer model generalizes much better on longer sequences than the attention model. Our final transducer model outperforms our attention model on Switchboard 300h by over 6% relative WER.

View Research

Early Stage LM Integration Using Local and Global Log-Linear Combination

May 2020
Wilfried Michel | Ralf Schlüter | Hermann Ney

Sequence-to-sequence models with an implicit alignment mechanism (e.g. attention) are closing the performance gap towards traditional hybrid hidden Markov models (HMM) for the task of automatic speech recognition. One important factor to improve word error rate in both cases is the use of an external language model (LM) trained on large text-only corpora. Language model integration is straightforward with the clear separation of acoustic model and language model in classical HMM-based modeling. In contrast, multiple integration schemes have been proposed for attention models. In this work, we present a novel method for language model integration into implicit-alignment based sequence-to-sequence models. Log-linear model combination of acoustic and language model is performed with a per-token renormalization. This allows us to compute the full normalization term efficiently both in training and in testing. This is compared to a global renormalization scheme which is equivalent to applying shallow fusion in training. The proposed methods show good improvements over standard model combination (shallow fusion) on our state-of-the-art Librispeech system. Furthermore, the improvements are persistent even if the LM is exchanged for a more powerful one after training.

View Research

Robust Beam Search for Encoder-Decoder Attention Based Speech Recognition without Length Bias

May 2020
Wei Zhou | Ralf Schlüter | Hermann Ney

As one popular modeling approach for end-to-end speech recognition, attention-based encoder-decoder models are known to suffer the length bias and corresponding beam problem. Different approaches have been applied in simple beam search to ease the problem, most of which are heuristic-based and require considerable tuning. We show that heuristics are not proper modeling refinement, which results in severe performance degradation with largely increased beam sizes. We propose a novel beam search derived from reinterpreting the sequence posterior with an explicit length modeling. By applying the reinterpreted probability together with beam pruning, the obtained final probability leads to a robust model modification, which allows reliable comparison among output sequences of different lengths. Experimental verification on the LibriSpeech corpus shows that the proposed approach solves the length bias problem without heuristics or additional tuning effort. It provides robust decision making and consistently good performance under both small and very large beam sizes. Compared with the best results of the heuristic baseline, the proposed approach achieves the same WER on the 'clean' sets and 4% relative improvement on the 'other' sets. We also show that it is more efficient with the additional derived early stopping criterion.

View Research

LSTM Language Models for LVCSR in First-Pass Decoding and Lattice-Rescoring

July 2019
Eugen Beck | Wei Zhou | Ralf Schlüter | Hermann Ney

LSTM based language models are an important part of modern LVCSR systems as they significantly improve performance over traditional backoff language models. Incorporating them efficiently into decoding has been notoriously difficult. In this paper we present an approach based on a combination of one-pass decoding and lattice rescoring. We perform d...

View Research

Effective Cross-lingual Transfer of Neural Machine Translation Models without Shared Vocabularies

July 2019
Y. Kim, Y. Gao, and H. Ney

Transfer learning or multilingual model is essential for low-resource neural machine translation (NMT), but the applicability is limited to cognate languages by sharing their vocabularies. This paper shows effective techniques to transfer a pre-trained NMT model to a new, unrelated language without shared vocabularies. We relieve the vocabulary mismatch by using cross-lingual word embedding, train a more language-agnostic encoder by injecting artificial noises, and generate synthetic data easily from the pre-training data without back-translation.....

View Research

Learning Bilingual Sentence Embeddings via Autoencoding and Computing Similarities with a Multilayer Perceptron

June 2019
Yunsu Kim | Hendrik Rosendahl | Nick Rossenbach | Hermann Ney

We propose a novel model architecture and training algorithm to learn bilingual sentence embeddings from a combination of parallel and monolingual data. Our method connects autoencoding and neural machine translation to force the source and target sentence embeddings to share the same space without the help of a pivot language or an additional transformation....

View Research

Language Modeling with Deep Transformers

May 2019
Kazuki Irie | Albert Zeyer | Ralf Schlüter | Hermann Ney

We explore multi-layer autoregressive Transformer models in language modeling for speech recognition. We focus on two aspects. First, we revisit Transformer model configurations specifically for language modeling. We show that well configured Transformer models outperform our baseline models based on the shallow stack of LSTM recurrent neural network layers....

View Research

Analysis of Deep Clustering as Preprocessing for Automatic Speech Recognition of Sparsely Overlapping Speech

May 2019
Tobias Menne, Ralf Schlüter, Hermann Ney:

Significant performance degradation of automatic speech recognition (ASR) systems is observed when the audio signal contains cross-talk. One of the recently proposed approaches to solve the problem of multi-speaker ASR is the deep clustering (DPCL) approach. Combining DPCL with a state-of-the-art hybrid acoustic model, we obtain a word...

View Research

Weakly Supervised Learning with Multi-Stream CNN-LSTM-HMMs to Discover Sequential Parallelism in Sign Language Videos

April 2019
Oscar Koller | Necati Cihan Camgoz | Hermann Ney | Richard Bowden

In this work we present a new approach to the field of weakly supervised learning in the video domain. Our method is relevant to sequence learning problems which can be split up into sub-problems that occur in parallel. Here, we experiment with sign language data. The approach exploits sequence constraints within each independent stream and combines them ....

View Research
View More Academic Research
30-Year Leaders in Speech Technology
Find us on Social Media:
ABOUT APPTEK

AppTek provides an artificial intelligence and machine learning-based automatic speech recognition, machine translation and natural language understanding platform for organizations in a variety of markets, such as media and entertainment, call centers, government, enterprise business and others across the globe. Available via the cloud or on-premise, AppTek delivers the highest quality real-time streaming and batch speech technology solutions in the industry.   Featuring scientists and research engineers who are recognized amongst the best and most experienced in the world, the company’s solutions cover a wide array of languages, dialects, and channels.

SEARCH APPTEK.COM
Copyright 2021 AppTek    |    Privacy Policy      |       Terms of Service     |      Cookie Policy