Patent classifications
G10L17/00
SYSTEMS AND METHODS FOR VIRTUAL MEETING SPEAKER SEPARATION
A computer-implemented machine learning method for improving speaker separation is provided. The method comprises processing audio data to generate prepared audio data and determining feature data and speaker data from the prepared audio data through a clustering iteration to generate an audio file. The method further comprises re-segmenting the audio file to generate a speaker segment and causing to display the speaker segment through a client device.
System and method for speaker recognition on mobile devices
A speaker recognition system for authenticating a mobile device user includes an enrollment and learning software module, a voice biometric authentication software module, and a secure software application. Upon request by a user of the mobile device, the enrollment and learning software module displays text prompts to the user, receives speech utterances from the user, and produces a voice biometric print. The enrollment and training software module determines when a voice biometric print has met at least a quality threshold before storing it on the mobile device. The secure software application prompts a user requiring authentication to repeat an utterance based at least on an attribute of a selected voice biometric print, receives a corresponding utterance, requests the voice biometric authentication software module to verify the identity of the second user using the utterance, and, if the user is authenticated, imports the voice biometric print.
System and method for speaker recognition on mobile devices
A speaker recognition system for authenticating a mobile device user includes an enrollment and learning software module, a voice biometric authentication software module, and a secure software application. Upon request by a user of the mobile device, the enrollment and learning software module displays text prompts to the user, receives speech utterances from the user, and produces a voice biometric print. The enrollment and training software module determines when a voice biometric print has met at least a quality threshold before storing it on the mobile device. The secure software application prompts a user requiring authentication to repeat an utterance based at least on an attribute of a selected voice biometric print, receives a corresponding utterance, requests the voice biometric authentication software module to verify the identity of the second user using the utterance, and, if the user is authenticated, imports the voice biometric print.
Computer systems exhibiting improved computer speed and transcription accuracy of automatic speech transcription (AST) based on a multiple speech-to-text engines and methods of use thereof
In some embodiments, an exemplary inventive system for improving computer speed and accuracy of automatic speech transcription includes at least components of: a computer processor configured to perform: generating a recognition model specification for a plurality of distinct speech-to-text transcription engines; where each distinct speech-to-text transcription engine corresponds to a respective distinct speech recognition model; receiving at least one audio recording representing a speech of a person; segmenting the audio recording into a plurality of audio segments; determining a respective distinct speech-to-text transcription engine to transcribe a respective audio segment; receiving, from the respective transcription engine, a hypothesis for the respective audio segment; accepting the hypothesis to remove a need to submit the respective audio segment to another distinct speech-to-text transcription engine, resulting in the improved computer speed and the accuracy of automatic speech transcription and generating a transcript of the audio recording from respective accepted hypotheses for the plurality of audio segments.
Computer systems exhibiting improved computer speed and transcription accuracy of automatic speech transcription (AST) based on a multiple speech-to-text engines and methods of use thereof
In some embodiments, an exemplary inventive system for improving computer speed and accuracy of automatic speech transcription includes at least components of: a computer processor configured to perform: generating a recognition model specification for a plurality of distinct speech-to-text transcription engines; where each distinct speech-to-text transcription engine corresponds to a respective distinct speech recognition model; receiving at least one audio recording representing a speech of a person; segmenting the audio recording into a plurality of audio segments; determining a respective distinct speech-to-text transcription engine to transcribe a respective audio segment; receiving, from the respective transcription engine, a hypothesis for the respective audio segment; accepting the hypothesis to remove a need to submit the respective audio segment to another distinct speech-to-text transcription engine, resulting in the improved computer speed and the accuracy of automatic speech transcription and generating a transcript of the audio recording from respective accepted hypotheses for the plurality of audio segments.
Speaker diartzation using an end-to-end model
Techniques are described for training and/or utilizing an end-to-end speaker diarization model. In various implementations, the model is a recurrent neural network (RNN) model, such as an RNN model that includes at least one memory layer, such as a long short-term memory (LSTM) layer. Audio features of audio data can be applied as input to an end-to-end speaker diarization model trained according to implementations disclosed herein, and the model utilized to process the audio features to generate, as direct output over the model, speaker diarization results. Further, the end-to-end speaker diarization model can be a sequence-to-sequence model, where the sequence can have variable length. Accordingly, the model can be utilized to generate speaker diarization results for any of various length audio segments.
Information processing device, information processing system, and information processing method, and program
Provided is a device, a method that allow a remote terminal to perform a process on the basis of a local-terminal-side user utterance. There are a local terminal and a remote terminal. The local terminal performs a process of a semantic analysis of a user utterance input into the local terminal. On the basis of a result of the semantic analysis, the local terminal determines whether or not the user utterance is a request to the remote terminal for a process. Moreover, in a case where the user utterance is a request to the remote terminal for a process, the local terminal transmits the result of the semantic analysis by a semantic-analysis part to the remote terminal. The remote terminal receives the result of the semantic analysis of the local-terminal-side user utterance, and performs a process based on the received result of the semantic analysis of the local-terminal-side user utterance.
System and method for simultaneous Layer 3 resiliency during audio capturing
The disclosed invention provide system and method to ensure resiliency in a network where audio capturing service experiences a failure on one or more active nodes. The network failover system is coupled to a Layer 3 (L3) network and communicates with a network switch through which network packets are transmitted. The failover system performs operations that include receiving network packets that are mirrored via the network switch, monitoring a primary node that captures audio data in the network packets, sending the network packets to a fallback node during an outage of the primary node, examining the network packets to determine which packets are audio-related packets, collecting audio-related packets, and storing the collected audio-related packets in a data storage. The fallback node is in the Layer 3 (L3) network.
System and method for simultaneous Layer 3 resiliency during audio capturing
The disclosed invention provide system and method to ensure resiliency in a network where audio capturing service experiences a failure on one or more active nodes. The network failover system is coupled to a Layer 3 (L3) network and communicates with a network switch through which network packets are transmitted. The failover system performs operations that include receiving network packets that are mirrored via the network switch, monitoring a primary node that captures audio data in the network packets, sending the network packets to a fallback node during an outage of the primary node, examining the network packets to determine which packets are audio-related packets, collecting audio-related packets, and storing the collected audio-related packets in a data storage. The fallback node is in the Layer 3 (L3) network.
Automated meeting minutes generation service
Attributes of electronic content from a meeting are identified and evaluated to determine whether sub-portions of the electronic content should or should not be attributed to a user profile. Upon determining that the sub-portion should be attributed to a user profile, attributes of the sub-portion of electronic content are compared to attributes of stored user profiles. A probability that the sub-portion corresponds to at least one stored user profile is calculated. Based on the calculated probability, the sub-portion is attributed to a stored user profile or a guest user profile.