1109 results about "Loudness" patented technology

Various dynamic audio ducking techniques are provided that may be applied where multiple audio streams, such as a primary audio stream and a secondary audio stream, are being played back simultaneously. For example, a secondary audio stream may include a voice announcement of one or more pieces of information pertaining to the primary audio stream, such as the name of the track or the name of the artist. In one embodiment, the primary audio data and the voice feedback data are initially analyzed to determine a loudness value. Based on their respective loudness values, the primary audio stream may be ducked during the period of simultaneous playback such that a relative loudness difference is generally maintained with respect to the loudness of the primary and secondary audio streams. Accordingly, the amount of ducking applied may be customized for each piece of audio data depending on its loudness characteristics.

Various dynamic audio ducking techniques are provided that may be applied where multiple audio streams, such as a primary audio stream and a secondary audio stream, are being played back simultaneously. For example, a secondary audio stream may include a voice announcement of one or more pieces of information pertaining to the primary audio stream, such as the name of the track or the name of the artist. In one embodiment, the primary audio data and the voice feedback data are initially analyzed to determine a loudness value. Based on their respective loudness values, the primary audio stream may be ducked during the period of simultaneous playback such that a relative loudness difference is generally maintained with respect to the loudness of the primary and secondary audio streams. Accordingly, the amount of ducking applied may be customized for each piece of audio data depending on its loudness characteristics.

An audio normalization gain value is applied to an audio signal to produce a normalized signal. The normalized signal is processed to compute dynamic range control (DRC) gain values in accordance with a selected one of several pre-defined DRC characteristics. The audio signal is encoded, and the DRC gain values are provided as metadata associated with the encoded audio signal. Several other embodiments are also described and claimed.

An indication of the loudness of an audio signal containing speech and other types of audio material is obtained by classifying segments of audio information as either speech or non-speech. The loudness of the speech segments is estimated and this estimate is used to derive the indication of loudness. The indication of loudness may be used to control audio signal levels so that variations in loudness of speech between different programs is reduced. A preferred method for classifying speech segments is described.

A method and an apparatus of recognizing whisper are provided. The method of recognizing a whisper may include recognizing a whispering action performed by a user through a first sensor, recognizing a loudness change through a second sensor, and activating a whisper recognition mode based on the whispering action and the loudness change.

A beeper system that includes a beeper attached to a flexible band. The flexible band is attached to the child, such as around the wrist, waist or neck, to prevent the child from removing the beeper. The beeper is programmed to reach a no shut off mode wherein a speaker outputs an audible output of increasing loudness until the beeper is turned off by keyswitch controlled or kept by the supervising adult. The keyswitch is also used to disconnect the flexible band to allow removal of the beeper. A wearer silence button is provided to allow for a warning period before the no shut off mode is reached.

A method for fitting a hearing compensation device comprises selecting a plurality of loudness levels for a plurality of frequencies and comparing each loudness level for each frequency for perceived sameness. The loudness levels may then be adjusted as needed to achieve perceived sameness across the frequency spectrum. A gain curve for each frequency is calculated from the selected plurality of loudness levels.

Embodiments disclosed herein enable detection and improvement of the quality of the audio signal using a mobile device by determining the loss in the audio signal and enhancing audio by streaming the remainder portion of audio. Embodiments disclosed herein enable an improvement in the sound quality rendered by rendering devices by emitting an test audio signal from the source device, measuring the test audio signal using microphones, detecting variation in the frequency response, loudness and timing characteristics using impulse responses and correcting for them. Embodiments disclosed herein also compensate for the noise in the acoustic space by determining the reverberation and ambient noise levels and their frequency characteristics and changing the digital filters and volumes of the source signal to compensate for the varying noise levels.

Methods and apparatus are disclosed for predicting events using acoustic and visual cues. The present invention processes audio and video information to identify one or more (i) acoustic cues, such as intonation patterns, pitch and loudness, (ii) visual cues, such as gaze, facial pose, body postures, hand gestures and facial expressions, or (iii) a combination of the foregoing, that are typically associated with an event, such as behavior exhibited by a video conference participant before he or she speaks. In this manner, the present invention allows the video processing system to predict events, such as the identity of the next speaker. The predictive speaker identifier operates in a learning mode to learn the characteristic profile of each participant in terms of the concept that the participant “will speak” or “will not speak” under the presence or absence of one or more predefined visual or acoustic cues. The predictive speaker identifier operates in a predictive mode to compare the learned characteristics embodied in the characteristic profile to the audio and video information and thereby predict the next speaker.

One or a combination of two or more specific loudness model functions selected from a group of two or more of such functions are employed in calculating the perceptual loudness of an audio signal. The function or functions may be selected, for example, by a measure of the degree to which the audio signal is narrowband or wideband. Alternatively or with such a selection from a group of functions, a gain value G[t] is calculated, which gain, when applied to the audio signal, results in a perceived loudness substantially the same as a reference loudness. The gain calculating employs an iterative processing loop that includes the perceptual loudness calculation.

A method for improving the intelligibility of an incoming telephone signal, including boosting loudness of at least one band of poorly heard frequencies of the signal within at least one band of intensities of the signal, the band lying below a predetermined intensity level at which telephone standard conformance testing is performed, thereby to generate a differentially boosted telephone signal. Alternatively or in addition, intelligibility of sibilants in a narrow band telephone signal is enhanced, by doubling the sampling rate of the narrow band signal by interpolation, thereby to provide a narrow band interpolated signal, generating a harmonic extrapolation signal by harmonically extrapolating from the narrow band interpolated signal thereby to estimate the missing portions of the telephone signal, the harmonic extrapolation comprising a sequence of pulses located at peaks of the interpolated signal, generating a missing energy estimator measure estimating energy missing at high frequency bands of the telephone signal, continuously modulating the amplitude of the pulses in said sequence of pulses based on said missing energy estimator measure, thereby to generate a modulated signal, passing the modulated signal through a shaping filter thereby to obtain a shaped signal, and summing the shaped signal with the interpolated signal.

A sound processing method for auditory prostheses, such as cochlear implants, which is adapted to improve the perception of loudness by users, and to improve speech perception. The overall contribution of stimuli to simulated loudness is compared with an estimate of acoustic loudness for a normally hearing listener based on the input sound signal. A weighting is applied to the filter channels to emphasize those frequencies which are most important to speech perception for normal hearing listeners when selecting channels as a basis for stimulation.

Integrated processing of multimedia signals can eliminate unnecessary signal processors and converters without losing the functionality of typical home entertainment system components. The integrated multimedia system includes a main player that captures and processes signals digitally. The main player may adjust the audio signal to provide audio output of equal loudness across all frequencies by accounting for sensitivity of the human ear for sounds of varying frequencies. The main player can also account for perceived differences in loudness based on the angle of a listener to a speaker by detecting the position of a user and making an adjustment accordingly. The invention further provides a speaker that has embedded performance characteristics or an identifier that allows the system to provide an optimal speaker driving current for a particular system or determine how that speaker would be best implemented in the integrated system.

An example of a method of correcting an audio level of a stored program asset comprises retrieving a stored program asset having audio encoded at a first loudness setting. Dialog of the audio of the asset is identified, a loudness of the dialog is determined and the determined loudness is compared to the first loudness setting. The asset is re-encoded at a second loudness setting corresponding to the determined loudness, if the first loudness setting and the second loudness are different by more than a predetermined amount. The determined loudness is preferably a DIALNORM of the dialog. The asset may be stored with the re-encoded loudness setting. The method may be applied to programs as they are being received from a source, as well. Aspects of the method may also be applied to programs to be provided by a source. Systems are also disclosed.

Transfer functions like Head Related Transfer Functions (HRTF) needed for binaural rendering are implemented efficiently by a subband-domain filter structure. In one implementation, amplitude, fractional-sample delay and phase-correction filters are arranged in cascade with one another and applied to subband signals that represent spectral content of an audio signal in frequency subbands. Other filter structures are also disclosed. These filter structures may be used advantageously in a variety of signal processing applications. A few examples of audio applications include signal bandwidth compression, loudness equalization, room acoustics correction and assisted listening for individuals with hearing impairments.

A method for fitting a hearing compensation device comprises selecting a plurality of loudness levels for a plurality of frequencies and comparing each loudness level for each frequency for perceived sameness. The loudness levels may then be adjusted as needed to achieve perceived sameness across the frequency spectrum. A gain curve for each frequency is calculated from the selected plurality of loudness levels.

A speech filter (108) enhances the loudness of a speech signal by expanding the formant regions of the speech signal beyond a natural bandwidth of the formant regions. The energy level of the speech signal is maintained so that the filtered speech signal contains the same energy as the pre-filtered signal. By expanding the formant regions of the speech signal on a critical band scale corresponding to human hearing, the listener of the speech signal perceives it to be louder even though the signal contains the same energy.

Embodiments are directed a method of rendering object-based audio comprising determining an initial spatial position of objects having object audio data and associated metadata, determining a perceptual importance of the objects, and grouping the audio objects into a number of clusters based on the determined perceptual importance of the objects, such that a spatial error caused by moving an object from an initial spatial position to a second spatial position in a cluster is minimized for objects with a relatively high perceptual importance. The perceptual importance is based at least in part by a partial loudness of an object and content semantics of the object.

A method and system are provided for equalizing the loudness of an audio source. Initially, the perceptual loudness level of an audio signal is measured from one or more audio sources. Next, the loudness level of the audio signal is adjusted using the perceptual loudness level. Thereafter, the audio signal corresponding to the music selections is reproduced such that the perceived loudness to a listener is the same entirely throughout a music track corresponding to the music selections.

The overall loudness of an audio track is calculated by combining a number of weighted loudness measures for segments of the audio track, where the weight applied to each individual loudness measure is a function of the loudness measure. By comparing the original overall loudness measure to a desired overall loudness measure, a gain can be determined that will adjust the loudness level to the desired value. Also disclosed is a dynamic compression method that analyzes the dynamic characteristics of an audio track and determines appropriate compressor parameters. Additionally, the loudness of a post-compressor audio track can be estimated for any given compressor parameters, thus permitting post-compression loudness matching to be done even if the compression is performed in real-time.

For measuring the influence of noise on the talking quality of a telephone link in a telecommunications network, a talker speech signal (s(t)) and a degraded speech signal (s'(t)) are fed to an objective measurement device (22) for obtaining an output signal (q) representing an estimated value of the talking quality. The degraded signal includes a returned signal (r(t)) originating from the network during transmission of the talker speech signal over the telephone link. The objective measurement carried out by the device is a modified PSQM-like measurement, which is modified as to include a modelling (32b) of masking effects in consequence of noise present in the returned signal. Preferably the modelling includes a noise suppression (42) carried out to a difference signal (D(t, f)) in the loudness density domain using a noise estimation (41).

In one disclosed aspect, dynamic gain modifications are applied to an audio signal at least partly in response to auditory events and / or the degree of change in signal characteristics associated with said auditory event boundaries. In another aspect, an audio signal is divided into auditory events by comparing the difference in specific loudness between successive time blocks of the audio signal.

Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.

Coding of received audio signals and the resulting application of electrical stimuli applied to electrodes used in a cochlear implant system are disclosed together with a method of fitting this new coding strategy. One of the aims is to place specific stimulation representing pitch by applying near threshold electrical stimuli with limited and focussed excitation fields. A range of stimulation rates and a minimal range of current levels above threshold are used for creation of a dynamic loudness percept for a cochlear implant recipient. Another aim is to disclose a coding scheme based on a model of physiological measures (i.e. refractoriness, adaptation, spread of activation field, spatiotemporal acoustical cochlear activation patterns and spontaneous activity) to estimate the proportions of available excitable auditory neurons close to the electrodes available for stimulation. The spectral bands formed from the pre-processing of incoming audio signals are weighted by these proportions of excitability to control place, timing, rate and current level of electrical stimuli applied to the electrodes available in the array.

A multiband DRC system and a method for controlling the same are disclosed. In accordance with the present invention, the system and the method are capable of increasing an overall loudness of an output signal by controlling thresholds and gains of plurality of DRCs included in the multiband DRC system according to frequency bands.

The invention relates to a method for objectively and quantifiably evaluating noise fret degree in a vehicle based on an auditory model, in particular to an evaluation method of psychoacoustics and vehicle sound quality. The method comprises the following steps: designing a dummy head model; imitating a processing mechanism of the middle ear and the inner ear of a human body to the sound by usingan auditory peripheral calculation model; collecting vehicle noise signals under the working conditions of uniform, accelerated and idle speed; pre-treating the noise sample and processing the loudness in specification; obtaining a psychoacoustics parameter by using a sound quality calculation model; obtaining subjective evaluation result test data by using a grouped and paired comparison method;calculating and painting a correlated scatter plot chart between each parameter and the ranking value of the subjective evaluation result; and analyzing and calculating to obtain the objectively quantifiable model of the subjective fret degree under each working condition. The invention can realize the psychoacoustic parameter calculation of objectively evaluating the sound quality with differentvehicle types, gears and speeds, wherein the calculated value has good pertinence and consistency with the evaluated result of the subjective evaluation method. The invention has stable evaluated result and high reliability, and can improve the sound quality and competitiveness of vehicles combined with the design of new CAE cars.