Abstract: An integrated circuit for digital signal routing. The integrated circuit has analog and digital inputs and outputs, including digital interfaces for connection to other integrated circuits. Inputs, including the digital interfaces, act as data sources. Outputs, including the digital interfaces, act as data destinations. The integrated circuit also includes signal processing blocks, which can act as data sources and data destinations. Signal routing is achieved by means of a multiply-accumulate block, which takes data from one or more data source and, after any required scaling, generates output data for a data destination. Data from a data source is buffered for an entire period of a data sample clock so that the multiply-accumulate block can retrieve the data at any point in the period, and output data of the multiply-accumulate block is buffered for an entire period of the data sample clock so that the data destination can retrieve the data at any point in the period.

Abstract: A multi-channel sound (MCS) system features intelligent calibration (e.g., of acoustic echo cancellation (AEC)) for use in dynamic acoustic environments. A sensor subsystem is utilized to detect and identify changes in the acoustic environment and determine a “scene” corresponding to the resulting acoustic characteristics for that environment. This detected scene is compared to predetermined scenes corresponding to the acoustic environment. Each predetermined scene has a corresponding pre-tuned filter configuration for optimal AEC performance. Based on the results of the comparison, the pre-tuned filter configuration corresponding to the predetermined scene that most closely matches the detected scene is utilized by the AEC subsystem of the multi-channel sound system.

Abstract: The present disclosure illustrates an acoustic echo cancellation method and system using the same. The acoustic echo cancellation method comprises the following steps. Firstly, a prior-knowledge matrix comprising a plurality of space vectors is built. Then, an initial filter vector is generated by the prior-knowledge matrix and an initial weighting vector. The weighting vector is updated based on the difference of the echo signal and the estimated signal in an iteration algorithm, and the coefficient of the filter vector is updated according to the updated weighting vector. An estimated signal is generated according to the updated filter vector and the original signal. Finally, the next echo signal is cancelled by the near-end estimated signal.

Abstract: When an indication is received from a user input, a name of an audio source from an ordered list of audio sources to which a headset is currently connected is output. Whenever a successive indication is received within a predefined amount of time, the name of the next wireless audio source in the list is output. If the next wireless audio source in the list is the last wireless audio source in the list, and the successive indication from the user input is received before the elapsed time exceeds the predefined value, the name of the audio source to which the headset is currently connected is output as the next selected wireless audio source in the list. When an amount of time greater than the predefined value elapses without a successive indication from the user input, the last wireless audio source that had its name output is connected.

Abstract: An audio rendering device enhances voice audio such that audible voice is not overwhelmed by other aspects of the soundtrack. The device attenuates right and left channels in an audio stream in response to a detected voice component in the audio stream, and boosts the voice component in the audio stream based on the level of attenuation of the right and left channels. Voice components are distinguished from the non-voice components by separating center channel and mono information from the left, right and surround channels. Non-voice components are attenuated down towards a non-voice threshold level based on an attenuation ratio. Voice components are boosted up toward a voice threshold level, so that the spoken voice is more audible to viewers and not overwhelmed or drowned out by the non-voice aspects of the soundtrack.

Abstract: An assistant guiding system includes a plurality of radio frequency identification (RFID) tags, a server, and an earphone assembly. The plurality of RFID tags is located at a predetermined area, and each RFID tag pre-stores an identification number of the predetermined area. The server stores scenic spot introduction information of the predetermined area. The earphone assembly communicates with a portable electronic device of tourists, and the portable electronic device downloads the scenic spot introduction information from the server. The earphone assembly includes a reading module and a transmission module. The reading module reads the identification number from the RFID tag. The transmission module sends the identification number to the portable electronic device. The scenic spot introduction information corresponding to the identification number of the predetermined area is played or displayed on the portable electronic device.

Abstract: A reverberator for reverberating an audio signal includes a feedback delay loop processor for delaying at least two frequency subband signals representing the audio signal by different loop delays to obtain reverberated frequency subband signals.

Abstract: Methods, apparatus, and computer-readable media are described herein related to using self-generated sounds for determining a worn state of a wearable computing device. A wearable computing device can transmit an audio signal. One or more sensors coupled to the wearable computing device may then receive a modified version of the audio signal. A comparison may be made between the modified version of the audio signal and at least one reference signal, where the at least one reference signal is based on the audio signal that is transmitted. Based on an output of the comparison, a determination can be made of whether the wearable computing device is being worn.

Abstract: An adaptive noise canceling (ANC) circuit adaptively generates an anti-noise signal from a reference microphone signal that is injected into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone proximate the speaker provides an error signal. A secondary path estimating adaptive filter estimates the electro-acoustical path from the noise canceling circuit through the transducer so that source audio can be removed from the error signal. Tones in the source audio, such as remote ringtones, present in downlink audio during initiation of a telephone call, are detected by a tone detector using accumulated tone persistence and non-silence hangover counting, and adaptation of the secondary path estimating adaptive filter is halted to prevent adapting to the tones. Adaptation of the adaptive filters is then sequenced so any disruption of the secondary path adaptive filter response is removed before allowing the anti-noise generating filter to adapt.

Abstract: A system capable of self-adjusting both sound level and spectral content to improve audibility and intelligibility of medical device audible cues. Audible cues are stored as sound files. Ambient noise is detected, and the output of the audible cues is altered based on the ambient noise. Various embodiments include processed sound files that are more robust in noisy environments.

Abstract: An information processing apparatus includes: an acquisition unit adapted to acquire activities corresponding to a plurality of bases from an audio signal; a determination unit adapted to determine whether the plurality of acquired activities have a specific characteristic; and an output unit adapted to output an audio signal obtained by separating, from the audio signal, a component corresponding to an activity which is determined to have the specific characteristic by the determination unit.

Abstract: An audio separating apparatus obtains a matrix by performing time-frequency conversion on an input audio signal. The audio separating apparatus divides the obtained matrix into at least a basis matrix and an activity matrix, and classifies base spectra that configure the respective columns of the basis matrix into first base spectra corresponding to a target sound and second base spectra corresponding to a non-target sound.

Abstract: A mobile terminal having a terminal body, an audio output module located in the terminal body, a power supply unit located in the terminal body, an earphone port formed on one side of the terminal body and a controller configured to, when a plug of earphone having a cable-type battery is inserted into the earphone port while performing an audio output function, to select an audio output path for at least one of the audio output module or the earphone inserted in the earphone port to output audio sound according to the audio output function and to charge the power supply unit by the cable-type battery is provided. Accordingly, an audio sound can be output through an audio output path selected according to the intention of the user who has inserted the earphone having the cable-type battery.

Abstract: In a multi-speaker audio system for, e.g., a home entertainment system or other entertainment system, each networked-speaker (wired or wireless) is activated or deactivated as appropriate such that audio play follows a user as the user moves around the home.

Abstract: An audio channel control circuit includes a filter module and a determining module. The audio channel control circuit receives a first audio signal and a second audio signal from an audio device. The first audio signal loads a first carrier signal, while the second audio signal loads a second carrier signal. The first carrier signal and the second carrier signal have different phases. The filter module filters the first audio signal and the second audio signal, separating the first carrier signal from the first audio signal and the second carrier signal from the second audio signal. According to the phases of the first carrier signal and the second carrier signal, the determining module chooses a speaker of the left audio channel or a speaker of the right audio channel to output the first audio signal or the second audio signal.

Abstract: Embodiments are provided for satellite volume control. An example method includes receiving an input at a playback device to adjust a volume for a plurality of playback devices that are grouped for synchronous playback of audio content, wherein the plurality of playback devices includes the playback device. The method also includes sending a first message over a network from the playback device to a device associated with the plurality of playback devices, the first message including information based on the input, wherein the information is used to adjust the volume of the plurality of playback devices. The method also includes receiving a second message at the playback device over the network, the second message including information for the volume of the playback device, wherein the volume is based on the adjusted volume of the plurality of playback devices.

Abstract: An example playback device includes one or more processors and a non-transitory computer-readable medium storing instructions that, when executed by the one or more processors, cause the playback device to perform functions. The functions include receiving digital data representing audio content. The functions further include sending at least a portion of the digital data to an identification system configured to identify the audio content based on the at least a portion of the digital data. The functions further include receiving information associated with the audio content from the identification system. The functions further include, in response to receiving the information associated with the audio content from the identification system, sending the received information to a control device that is configured to control the playback device. The functions further include playing the audio content via one or more speakers electrically connected to the playback device via a wired connection.

Abstract: A bionic hearing headset for enhancing directional sound from an external audio source. The headset includes a pair of headphones, each having a microphone array that connects listeners to the environment through a plurality of microphones, even while listening to content presented over the headphones from an electronic audio source. The microphone array signals are first converted into beam-formed directional signals. Diffuse signal components may be suppressed using a common, noise-reduction mask. The audio signals may then be converted to binaural format using a plurality of head-related transfer function (HRTF) pairs.

Abstract: Provided are methods and systems for concealing missing segments and/or discontinuities in an audio signal, thereby restoring the continuity of the signal. The methods and systems are designed for and targeted at audio signals, are based on interpolation and extrapolation operations for sinusoids, and do not rely on the assumption that the sinusoids are harmonic. The methods and systems are improvements over existing audio concealment approaches in that, among other advantages, the methods and systems facilitate asynchronous interpolation, use an interpolation procedure that corresponds to time-domain waveform interpolation if the signal is harmonic, and have a peak selection procedure that is effective for audio signals.

Abstract: A method for feedback detection by an electronic device is described. The method includes receiving a first microphone signal by a first microphone. A feedback loop includes the first microphone and a speaker. The method also includes receiving a second microphone signal by a second microphone that is outside of the feedback loop. A first signal based on the first microphone signal and a second signal based on the second microphone signal exhibit a higher correlation in presence of feedback and exhibit a lower correlation in absence of feedback. The method further includes determining a correlation based on the first microphone signal and the second microphone signal. The method additionally includes determining whether feedback is occurring based on the correlation.