Abstract:
Techniques for speech processing using a deep neural network (DNN) based acoustic model front-end are described. A new modeling approach directly models multi-channel audio data received from a microphone array using a first model (e.g., multi-channel DNN) that takes in raw signals and produces a first feature vector that may be used similarly to beamformed features generated by an acoustic beamformer. A second model (e.g., feature extraction DNN) processes the first feature vector and transforms it to a second feature vector having a lower dimensional representation. A third model (e.g., classification DNN) processes the second feature vector to perform acoustic unit classification and generate text data. These three models may be jointly optimized for speech processing (as opposed to individually optimized for signal enhancement), enabling improved performance despite a reduction in microphones and a reduction in bandwidth consumption during real-time processing.
Abstract:
Features are disclosed for modeling user interaction with a detection system using a stochastic dynamical model in order to determine or adjust detection thresholds. The model may incorporate numerous features, such as the probability of false rejection and false acceptance of a user utterance and the cost associated with each potential action. The model may determine or adjust detection thresholds so as to minimize the occurrence of false acceptances and false rejections while preserving other desirable characteristics. The model may further incorporate background and speaker statistics. Adjustments to the model or other operation parameters can be implemented based on the model, user statistics, and/or additional data.
Abstract:
Features are disclosed for spotting keywords in utterance audio data without requiring the entire utterance to first be processed. Likelihoods that a portion of the utterance audio data corresponds to the keyword may be compared to likelihoods that the portion corresponds to background audio (e.g., general speech and/or non-speech sounds). The difference in the likelihoods may be determined, and keyword may be triggered when the difference exceeds a threshold, or shortly thereafter. Traceback information and other data may be stored during the process so that a second speech processing pass may be performed. For efficient management of system memory, traceback information may only be stored for those frames that may encompass a keyword; the traceback information for older frames may be overwritten by traceback information for newer frames.
Abstract:
Approaches are described for detecting when an electronic device (such as a mobile phone) has been stolen or is otherwise being used by someone other than an authorized user of the device. At least one sensor of the device can obtain data during a current use of the device, and the device can determine from the data a set of available features. The features can be compared to a corresponding model associated with an owner (or other authorized user) of the device to generate a confidence value indicative of whether the current user operating the device is likely the owner of the device. The confidence value can be compared to at least one confidence threshold, for example, and based on the comparison, the current user can be provided access to at least a portion of functionality of the device and/or a security action can be performed when the confidence value does not at least meet at least one confidence threshold.
Abstract:
Features are disclosed for detecting words in audio using environmental information and/or contextual information in addition to acoustic features associated with the words to be detected. A detection model can be generated and used to determine whether a particular word, such as a keyword or “wake word,” has been uttered. The detection model can operate on features derived from an audio signal, contextual information associated with generation of the audio signal, and the like. In some embodiments, the detection model can be customized for particular users or groups of users based usage patterns associated with the users.
Abstract:
A speech-based audio device may be configured to detect a user-uttered trigger expression and to respond by interpreting subsequent words or phrases as commands. In order to distinguish between utterance of the trigger expression by the user and generation of the trigger expression by the device itself, output signals used as speaker inputs are analyzed to detect whether the trigger expression has been generated by the speaker. If a detected trigger expression has been generated by the speaker, it is disqualified. Disqualified trigger expressions are not acted upon the by the audio device.
Abstract:
Features are disclosed for spotting keywords in utterance audio data without requiring the entire utterance to first be processed. Likelihoods that a portion of the utterance audio data corresponds to the keyword may be compared to likelihoods that the portion corresponds to background audio (e.g., general speech and/or non-speech sounds). The difference in the likelihoods may be determined, and keyword may be triggered when the difference exceeds a threshold, or shortly thereafter. Traceback information and other data may be stored during the process so that a second speech processing pass may be performed. For efficient management of system memory, traceback information may only be stored for those frames that may encompass a keyword; the traceback information for older frames may be overwritten by traceback information for newer frames.
Abstract:
An automatic speech recognition (ASR) system detects an endpoint of an utterance using the active hypotheses under consideration by a decoder. The ASR system calculates the amount of non-speech detected by a plurality of hypotheses and weights the non-speech duration by the probability of each hypotheses. When the aggregate weighted non-speech exceeds a threshold, an endpoint may be declared.
Abstract:
In an automatic speech recognition (ASR) processing system, ASR processing may be configured to process speech based on multiple channels of audio received from a beamformer. The ASR processing system may include a microphone array and the beamformer to output multiple channels of audio such that each channel isolates audio in a particular direction. The multichannel audio signals may include spoken utterances/speech from one or more speakers as well as undesired audio, such as noise from a household appliance. The ASR device may simultaneously perform speech recognition on the multi-channel audio to provide more accurate speech recognition results.
Abstract:
Features are disclosed for managing the use of speech recognition models and data in automated speech recognition systems. Models and data may be retrieved asynchronously and used as they are received or after an utterance is initially processed with more general or different models. Once received, the models and statistics can be cached. Statistics needed to update models and data may also be retrieved asynchronously so that it may be used to update the models and data as it becomes available. The updated models and data may be immediately used to re-process an utterance, or saved for use in processing subsequently received utterances. User interactions with the automated speech recognition system may be tracked in order to predict when a user is likely to utilize the system. Models and data may be pre-cached based on such predictions.