Abstract:
A dual mode speech recognition system sends speech to two or more speech recognizers. If a first recognition result is received, whose recognition score exceeds a high threshold, the first result is selected without waiting for another result. If the score is below a low threshold, the first result is ignored. At intermediate values of recognition scores, a timeout duration is dynamically determined as a function of the recognition score. The timeout duration determines how long the system will wait for another result. Many functions of the recognition score are possible, but timeout durations generally decrease as scores increase. When receiving a second recognition score before the timeout occurs, a comparison based on recognition scores determines whether the first result or the second result is the basis for creating a response.
Abstract:
Speech synthesis chooses pronunciations of words with multiple acceptable pronunciations based on an indication of a personal, class-based, or global preference or an intended non-preferred pronunciation. A speaker's words can be parroted back on personal devices using preferred pronunciations for accent training. Degrees of pronunciation error are computed and indicated to the user in a visual transcription or audibly as word emphasis in parroted speech. Systems can use sets of phonemes extended beyond those generally recognized for a language. Speakers are classified in order to choose specific phonetic dictionaries or adapt global ones. User profiles maintain lists of which pronunciations are preferred among ones acceptable for words with multiple recognized pronunciations. Systems use multiple correlations of word preferences across users to predict use preferences of unlisted words. Speaker-preferred pronunciations are used to weight the scores of transcription hypotheses based on phoneme sequence hypotheses in speech engines.
Abstract:
A system and method is presented for performing dual mode speech recognition, employing a local recognition module on a mobile device and a remote recognition engine on a server device. The system accepts a spoken query from a user, and both the local recognition module and the remote recognition engine perform speech recognition operations on the query, returning a transcription and confidence score, subject to a latency cutoff time. If both sources successfully transcribe the query, then the system accepts the result having the higher confidence score. If only one source succeeds, then that result is accepted. In either case, if the remote recognition engine does succeed in transcribing the query, then a client vocabulary is updated if the remote system result includes information not present in the client vocabulary.
Abstract:
In one implementation, a method is described of retrying matching of an audio query against audio references. The method includes receiving a follow-up query that requests a retry at matching a previously submitted audio query. In some implementations, this follow-up query is received without any recognition hint that suggests how to retry matching. The follow-up query includes the audio query or a reference to the audio query to be used in the retry. The method further includes retrying matching the audio query using retry matching resources that include an expanded group of audio references, identifying at least one match and transmitting a report of the match. Optionally, the method includes storing data that correlates the follow-up query, the audio query or the reference to the audio query, and the match after retrying.
Abstract:
A client, such as a mobile phone, receives an audio signal from a microphone; the sound comes from a broadcast signal such as a radio or television program. The client sends a segment of audio data from the broadcast program to a detection system, such as a server. A broadcast monitoring system receives many broadcast audio signals and encodes their fingerprints in a database for matching. The detection system compares the client's audio data fingerprints to the content fingerprints to identify which broadcast station broadcast the signal having the sampled content. This information enables the client to resume the experience of the broadcast from one of a number of possible media sources.
Abstract:
The present invention relates to the continuous monitoring of an audio signal and identification of audio items within an audio signal. The technology disclosed utilizes predictive caching of fingerprints to improve efficiency. Fingerprints are cached for tracking an audio signal with known alignment and for watching an audio signal without known alignment, based on already identified fingerprints extracted from the audio signal. Software running on a smart phone or other battery-powered device cooperates with software running on an audio identification server.
Abstract:
Methods and systems for correction of a likely erroneous word in a speech transcription are disclosed. By evaluating token confidence scores of individual words or phrases, the automatic speech recognition system can replace a low-confidence score word with a substitute word or phrase. Among various approaches, neural network models can be used to generate individual confidence scores. Such word substitution can enable the speech recognition system to automatically detect and correct likely errors in transcription. Furthermore, the system can indicate the token confidence scores on a graphic user interface for labeling and dictionary enhancement.
Abstract:
A method of controlling an engagement state of an agent during a human-machine dialog is provided. The method can include receiving a spoken request that is a conditional locking request, wherein the conditional locking request uses a natural language expression to explicitly specify a locking condition, which is a predicate, storing the predicate in a format that can be evaluated when needed by the agent, entering a conditionally locked state in response to the conditional locking request, in the conditionally locked state, receiving a multiplicity of requests without a need for a wakeup indicator, and for a request from the multiplicity of requests evaluating the predicate upon receiving the request, and processing the request if the predicate is true.
Abstract:
A machine learning system for a digital assistant is described, together with a method of training such a system. The machine learning system is based on an encoder-decoder sequence-to-sequence neural network architecture trained to map input sequence data to output sequence data, where the input sequence data relates to an initial query and the output sequence data represents canonical data representation for the query. The method of training involves generating a training dataset for the machine learning system. The method involves clustering vector representations of the query data samples to generate canonical-query original-query pairs in training the machine learning system.
Abstract:
A server supports multiple virtual assistants. It receives requests that include wake phrase audio and an identification of the source of the request, such as a virtual assistant device. Based on the identification, the server searches a database for a wake phrase detector appropriate for the identified source. The server then applies the wake phrase detector to the received wake phrase audio. If the wake phrase audio triggers the wake phrase detector, the server provides an appropriate response to the source.