A system capable of self-adjusting both sound level and spectral content to improve audibility and intelligibility of electronic 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.

A system capable of self-adjusting both sound level and spectral content to improve audibility and intelligibility of electronic 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.

There is disclosed a portable programmable device including a battery, a memory and a terminal connectable to earpieces, the device including in the memory a calibration file, parameter or parameters relating to audio sensitivity of the earpieces, the device being configured to play media data including audio, and to provide audio output to the earpieces, the device being further configured to, using the calibration file, parameter or parameters, calculate a noise dose relating to a sound exposure of a user resulting from audio output provided to the earpieces, and to record the noise dose on the device, wherein the device is configured to adjust audio output level in response to: (a) audio content included in played media data; (b) the calibration file, parameter or parameters, and (c) noise dose data of the user recorded on the device. A related method and computer program product are also disclosed.

Disclosed are examples of systems, apparatus, methods and computer-readable storage media for dynamically adjusting thresholds of a compressor. An input audio signal having a number of frequency band components is processed. Time-varying thresholds can be determined. A compressor performs, on each frequency band component, a compression operation having a corresponding time-varying threshold to produce gains. Each gain is applied to a delayed corresponding frequency band component to produce processed band components, which are summed to produce an output signal. In some implementations, a time-varying estimate of a perceived spectrum of the output signal and a time-varying estimate of a distortion spectrum induced by the perceived spectrum estimate are determined, for example, using a distortion audibility model. An audibility measure of the distortion spectrum estimate in the presence of the perceived spectrum estimate can be predicted and used to adjust the time-varying thresholds.

An example method includes receiving data indicating a configuration of one or more playback devices. The one or more playback devices may include one or more transducers. The method further includes, based on the received data, associating each of one or more audio streams respectively with at least one transducer of the one or more transducers. The method further includes generating the one or more audio streams and sending at least one of the generated one or more audio streams to each of the one or more playback devices. An example non-transitory computer readable medium and an example computing device related to the example method are also disclosed herein.

A speaker has a magnetic angle sensor that detects the displacement of the vibration system of the speaker. A non-linearity compensation filter compensates for output distortion due to a non-linear parameter of the speaker. A transfer function of a linear inverse filter is applied so that output distortion is eliminated from the displacement, detected by the magnetic angle sensor, of the vibration system. A controller drives the speaker by using a test signal, and measures a response from the detected displacement of the vibration system. If error between the measured response and the theoretical value, theoretically determined from the design specifications, of the response is small, the controller uses the theoretical value, theoretically determined from the design specifications, of the non-linear parameter as an active non-linear parameter and sets, in the non-linearity compensation filter, a transfer function that compensates for distortion in an output due to the active non-linear parameter.

A method of generating audio loudness performed by an audio loudness generation device may include: receiving loudness information on each of a plurality of audio tracks included in one group; predicting an intermediate loudness distribution, which is a loudness distribution for the one group, on the basis of the loudness information on each of the plurality of audio tracks; and generating an integrated loudness for the one group on the basis of the intermediate loudness distribution.

In some embodiments, a method for performing enhancement on an audio signal to generate an enhanced audio signal in response to feedback indicative of amount of compression applied to at least one frequency band of the enhanced audio signal. In typical embodiments, the enhancement is or includes bass enhancement. Examples of other types of enhancement performed in other embodiments include dialog enhancement, upmixing, frequency shifting, harmonic injection or transposition, subharmonic injection, virtualization, and equalization. Other aspects are systems (e.g., programmed processors) and devices (e.g., devices having physically-limited bass reproduction capabilities, such as, for example, a notebook, tablet, mobile phone, or other device with small speakers) configured to perform any embodiment of the method.

The invention relates to a hearing aid a cochlear implant comprising a) at least one input transducer for capturing incoming sound and for generating electric audio signals which represent frequency bands of the incoming sound, b) a sound processor which is configured to analyze and to process the electric audio signals, c) a transmitter that sends the processed electric audio signals, d) a receiver/stimulator, which receives the processed electric audio signals from the transmitter and converts the processed electric audio signals into electric pulses, e) an electrode array embedded in the cochlear comprising a number of electrodes for stimulating the cochlear nerve with said electric pulses, and f) a control unit configured to control the distribution of said electric pulses to the number of said electrodes. The control unit is configured to distribute said electric pulses to the number of said electrodes by applying one out of a plurality of different coding schemes, and wherein the applied coding scheme is selected according to characteristics of the incoming sound.

A system and method for optimizing the low-frequency sound rendition of an audio signal, implementing variations in a plurality of parameters of the audio signal according to the volume level of the signal chosen by a user, in particular filtering or compression parameters, or parameters relating to the harmonics of the audio signal, while seeking to optimize the dynamics and the bandwidth of the audio signal according to the volume, in order to provide an optimal rendition to the user.