摘要:
A method and device for performing voice control on a device with a microphone array are disclosed. The method includes the following steps. It is confirmed that the device is in an audio playing state. An interference sound interfering the device in the audio playing state is analyzed. A voice enhancement mode adopted by the device is selected according to a feature of the interference sound. A user's voice is detected in real time for a wake-up word, and when the wake-up word is detected, the device is controlled to stop audio playing. An interference sound interfering the device after playing audios is stopped is analyzed, and the voice enhancement mode adopted by the device is adjusted according to a feature of the interference sound. A command word from a user is acquired to control the device to execute a corresponding function, to respond to the user. The present solution can improve voice recognition accuracy of the equipment, and improve user experience.
摘要:
The present invention describes methods for performing large-scale graph traversal calculations on parallel processor platforms. The invention describes methods for on-the-fly hypothesis rescoring that utilizes graphic processing units (GPUs) in combination with utilizing central processing units (CPUs) of computing devices. The invention is described in one embodiment as applied to the task of large vocabulary continuous speech recognition.
摘要:
Method and system for real-time speech recognition is provided. The speech algorithm runs on a platform having an input-output processor and a plurality of processor units. The processor units operate substantially in parallel or sequentially to perform feature extraction and pattern matching. While the input-output processor creates a frame, the processor units execute the feature extraction and the pattern matching. Shared memory is provided for supporting the parallel operation.
摘要:
A system, article, and method of automatic speech recognition using parallel processing for weighted finite state transducer-based speech decoding.
摘要:
A feature transform for speech recognition is described. An input speech utterance is processed to produce a sequence of representative speech vectors. A single time-synchronous speech recognition pass is performed using a decoding search to determine a recognition output corresponding to the speech input. The decoding search includes, for each speech vector after some first threshold number of speech vectors, estimating a feature transform based on the preceding speech vectors in the utterance and partial decoding results of the decoding search. The current speech vector is then adjusted based on the current feature transform, and the adjusted speech vector is used in a current frame of the decoding search.
摘要:
Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for recognizing speech using a variable length of context. Speech data and data identifying a candidate transcription for the speech data are received. A phonetic representation for the candidate transcription is accessed. Multiple test sequences are extracted for a particular phone in the phonetic representation. Each of the multiple test sequences includes a different set of contextual phones surrounding the particular phone. Data indicating that an acoustic model includes data corresponding to one or more of the multiple test sequences is received. From among the one or more test sequences, the test sequence that includes the highest number of contextual phones is selected. A score for the candidate transcription is generated based on the data from the acoustic model that corresponds to the selected test sequence.
摘要:
In syllable or vowel or phone boundary detection during speech, an auditory spectrum may be determined for an input window of sound and one or more multi-scale features may be extracted from the auditory spectrum. Each multi-scale feature can be extracted using a separate two-dimensional spectro-temporal receptive filter. One or more feature maps corresponding to the one or more multi-scale features can be generated and an auditory gist vector can be extracted from each of the one or more feature maps. A cumulative gist vector may be obtained through augmentation of each auditory gist vector extracted from the one or more feature maps. One or more syllable or vowel or phone boundaries in the input window of sound can be detected by mapping the cumulative gist vector to one or more syllable or vowel or phone boundary characteristics using a machine learning algorithm.
摘要:
Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for recognizing speech using a variable length of context. Speech data and data identifying a candidate transcription for the speech data are received. A phonetic representation for the candidate transcription is accessed. Multiple test sequences are extracted for a particular phone in the phonetic representation. Each of the multiple test sequences includes a different set of contextual phones surrounding the particular phone. Data indicating that an acoustic model includes data corresponding to one or more of the multiple test sequences is received. From among the one or more test sequences, the test sequence that includes the highest number of contextual phones is selected. A score for the candidate transcription is generated based on the data from the acoustic model that corresponds to the selected test sequence.