Abstract: The present disclosure provides a speech recognition method and device. The method includes: receiving a speech signal; decoding the speech signal according to an acoustic model, a language model and a decoding network established in advance, and dynamically adding a blank unit in a decoding process to obtain an optimum decoding path with the added blank unit, in which the acoustic model is obtained based on connectionist temporal classification training, the acoustic model includes basic pronunciation units and the blank unit, and the decoding network includes a plurality of decoding paths consisting of the basic pronunciation units; and outputting the optimum decoding path as a recognition result of the speech signal.

Abstract: Disclosed is an apparatus and method for processing a multichannel audio signal. A multichannel audio signal processing method may include: generating an N-channel audio signal of N channels by down-mixing an M-channel audio signal of M channels; and generating a stereo audio signal by performing binaural rendering of the N-channel audio signal.

Abstract: Embodiments of the disclosure include an improved content streaming system that is configured to simplify and streamline the process of streaming media content from one or more content providers to one or more electronic devices. In some embodiments, the interaction of a user with one or more components in a content distribution system is used to initiate the streaming of media content to one or more content players from either a first content server or a second content server.

Abstract: The present disclosure describes a system and method for determining cardiac parameters and physiological conditions of a user by analysing speech samples of said user. A user device of the user may record specifics of speech and use these specifics of speech as a speech sample of user's utterance. The user device may transmit the speech samples to a backend system. The system may isolate phonation segments from the speech samples. The system may filter one or more phonation segments. The system may isolate uttered speech segments from one or more phonation segments. The system may perform an acoustic-phonetic analysis of the uttered speech segments. The acoustic-phonetic analysis may use plurality of features for the analysis. The IPA phonemes may be used to derive speech markers that correspond to specific cardiac parameters and physiological conditions. The system may generate a resulting report after analysis which is transmitted to the user.

Abstract: A technique determination device according to one embodiment of the present invention comprises an input sound acquisition unit acquiring an input sound, a pitch detection unit detecting a pitch on a time-series basis based on the input sound, a sound-volume detection unit detecting a sound volume on the time series basis based on the input sound, a first starting-point detection unit determining whether variation of the sound volume is equal to or larger than a predetermined threshold for each predetermined period and detecting a starting point of a period in which the variation of the sound volume is equal to or larger than the threshold as a first starting point, and a technique determination unit determining a technique of the input sound based on a change of the sound volume after the first starting point and variation of the pitch after the first starting point.

Abstract: Disclosed is an electronic device including: at least one processor; and a memory electrically connected to the at least one processor, wherein the memory stores instructions to recognize a received first voice, recognizes a first speaker based on the recognized first voice, and determines a response corresponding to the first voice based on a result of the recognition of the first speaker. Other embodiments are possible.

Abstract: Embodiments of the example embodiment relate to audio object extraction. A method for audio object extraction from audio content is disclosed. The method comprises determining a sub-band object probability for a sub-band of the audio signal in a frame of the audio content, the sub-band object probability indicating a probability of the sub-band of the audio signal containing an audio object. The method further comprises splitting the sub-band of the audio signal into an audio object portion and a residual audio portion based on the determined sub-band object probability. Corresponding system and computer program product are also disclosed.

Abstract: The positions of a plurality of speakers at a media consumption site are determined. Audio information in an object-based format is received. Gain adjustment value for a sound content portion in the object-based format may be determined based on the position of the sound content portion and the positions of the plurality of speakers. Audio information in a ring-based channel format is received. Gain adjustment value for each ring-based channel in a set of ring-based channels may be determined based on the ring to which the ring-based channel belongs and the positions of the speakers at a media consumption site.

Abstract: This disclosure describes, in part, techniques for implementing voice-enabled devices in vehicle environments to facilitate voice interaction with vehicle computing devices. Due to the differing communication capabilities of existing vehicle computing devices, the techniques described herein describe different communication topologies for facilitating voice interaction with the vehicle computing devices. In some examples, the voice-enabled device may be communicatively coupled to a user device, which may communicate with a remote speech-processing system to determine and perform operations responsive to the voice commands, such as conducting phone calls using loudspeakers of the vehicle computing device, streaming music to the vehicle computing device, and so forth.

Abstract: Method, device, and storage medium for generating and self-adaptively rotating a voice recognition interface are provided. The method for generating and self-adaptively rotating a voice recognition interface includes: identifying a direction of an interface layout of a current interface when a voice recognition is triggered; generating the voice recognition interface in a same layout direction as the direction of the current interface layout, the voice recognition interface covering a portion of a screen; monitoring in real time whether the interface layout of the current interface rotates; and rotating the generated voice recognition interface self-adaptively along with the current interface layout, in response to a rotation of the interface layout of the current interface.

Abstract: A mobile construction machine detects a speech processing trigger. It then performs speech processing (such as speech recognition and natural language understanding, speech synthesis, etc.) based on the detected speech processing trigger, to generate a speech processing result. A control signal generator generates control signals based on the speech processing result. The control signals can be used to control the mobile construction machine, to control another mobile construction machine, to provide information to a remote server location, or to aggregate information from multiple remote server locations.

Abstract: Embodiments of the present disclosure provide a method and a device for extracting a speech feature based on artificial intelligence. The method includes performing a spectrum analysis on a speech to be recognized, to obtain a spectrum program of the speech; and extracting features of the spectrum program by using an Inception convolution structure of an image recognition algorithm, to obtain the speech feature of the speech. In embodiments, by performing the spectrum analysis on the speech to be recognized, the consecutive speech to be recognized is converted into the spectrum diagram. As the Inception convolution structure is an effective image recognition manner being able to accurately recognize features of an image, the spectrum program is recognized with the Inception convolution structure to extract the relative accurate speech feature from the speech to be recognized. Thus, the accuracy rate of the speech recognition is improved.

Abstract: A microphone circuit having an amplifier with an input operably coupled to a microphone motor also includes a low pass filter operably coupled to the output of the amplifier and a positive feedback network that operably couples to an output of the low-pass filter and to the amplifier input. For many useful application settings the aforementioned amplifier has unity gain while the positive feedback network has a fractional gain less than unity.

Abstract: Embodiments of the disclosure include an improved content streaming system that is configured to simplify and streamline the process of streaming media content from one or more content providers to one or more electronic devices. In some embodiments, the interaction of a user with one or more components in a content distribution system is used to initiate the streaming of media content to one or more content players from either a first content server or a second content server.

Abstract: This disclosure relates to speakers and more specifically to an array speaker for distributing music uniformly across a room. A number of audio drivers can be radially distributed within a speaker housing so that an output of the drivers is distributed evenly throughout the room. In some embodiments, the exit geometry of the audio drivers can be configured to bounce off a surface supporting the array speaker to improve the distribution of music throughout the room. The array speaker can include a number of vibration isolation elements distributed within a housing of the array speaker. The vibration isolation elements can be configured to reduce the strength of forces generated by a subwoofer of the array speaker.

Abstract: A dynamic boost audio system includes a booster circuit having a dynamically sliding power supply unit (PSU) capable of outputting power among a plurality of different power levels. The booster circuit is configured to identify a real-time audio level of an audio signal, and automatically adjust the power to the power level such that the audio signal is output in response to the real-time audio level.

Abstract: The application relates to MEMS microphone transducers having a vent structure provided in the membrane layer and an opening formed at the vent structure for tuning the frequency response of the microphone.

Abstract: There are provided decoding and encoding methods for encoding and decoding of multichannel audio content for playback on a speaker configuration with N channels. The decoding method comprises decoding, in a first decoding module, M input audio signals into M mid signals which are suitable for playback on a speaker configuration with M channels; and for each of the N channels in excess of M channels, receiving an additional input audio signal corresponding to one of the M mid signals and decoding the input audio signal and its corresponding mid signal so as to generate a stereo signal including a first and a second audio signal which are suitable for playback on two of the N channels of the speaker configuration.

Abstract: Disclosed is an apparatus and method for processing a multichannel audio signal. A multichannel audio signal processing method may include: generating an N-channel audio signal of N channels by down-mixing an M-channel audio signal of M channels; and generating a stereo audio signal by performing binaural rendering of the N-channel audio signal.

Abstract: One or more embodiments set forth an audio processing system for a personal listening device that includes a set of microphones, a noise reduction module, an audio ducker, and a mixer. The set of microphones is configured to receive a first set of audio signals from an environment. The noise reduction module is configured to detect when a signal of interest is present in the first plurality of audio signals, and, upon detecting a signal of interest, transmit a ducking control signal. The audio ducker is configured to receive the ducking control signal, and receive a second plurality of audio signals via a playback device. The audio ducker is further configured to reduce an amplitude of a second plurality of audio signals relative to the signal of interest based on the ducking control signal. The mixer combines the first plurality of audio signals and second plurality of audio signals.