32 results about "Loudness compensation" patented technology

The invention relates to the measurement and control of the perceived sound loudness and / or the perceived spectral balance of an audio signal. An audio signal is modified in response to calculations performed at least in part in the perceptual (psychoacoustic) loudness domain. The invention is useful, for example, in one or more of: loudness-compensating volume control, automatic gain control, dynamic range control (including, for example, limiters, compressors, expanders, etc.), dynamic equalization, and compensating for background noise interference in an audio playback environment. The invention includes not only methods but also corresponding computer programs and apparatus.

Audio signal processing relating to the measurement and control of the perceived sound loudness and / or the perceived spectral balance of an audio signal is useful, for example, in one or more of: loudness-compensating volume control, automatic gain control, dynamic range control (including, for example, limiters, compressors, expanders, etc.), dynamic equalization, and compensating for background noise interference in an audio playback environment. In various embodiments, modification parameters are derived for modifying the audio signal in order to reduce the difference between its specific loudness and a target specific loudness.

A speech correction apparatus includes a speaker for generating guidance speech; a microphone set at a hearing position; an acoustic-characteristic setting unit for separating ambient noise from the guidance speech at the hearing position; an operating unit; a speech correcting filter for correcting the sound pressure level of the guidance speech generated by the speaker based on the average power of the guidance speech and the average power of the ambient noise which are separated; a loudness-compensating-gain calculating unit; and a speech-head correcting unit for correcting the average power of the guidance speech corresponding to the speech head at the border between a silent state and a speech state of the guidance speech.

A loudness-compensating volume control method imposes a desired loudness scaling on an audio signal by processing the audio signal in both the digital and analog domains by receiving a desired loudness scaling, deriving a wideband gain component and one or more other gain components from the desired loudness scaling, applying in the digital domain modifications to the audio signal based on the one or more other gain components to produce a partly-modified audio signal, and applying in the analog domain modifications to the partly-modified audio signal based on the wideband gain component. Additional loudness modifications other than volume control loudness modifications on the audio signal may also be imposed.

An audio correcting apparatus includes a speaker provided on a television apparatus, a microphone provided on a remote controller, an identifying unit which identifies an acoustic characteristic from the speaker to the microphone, and an acoustic characteristic setting unit having the acoustic characteristic. A signal obtained by allowing an audio signal input to the speaker to pass through the acoustic characteristic setting unit, and a signal representing ambient noise are input to an audio-correcting filter and a loudness-compensation-gain calculating unit. Based on both signals, the sound pressure level of sound output from the speaker is corrected so that the sound output from the speaker is clearly heard when reaching the user without being affected by the ambient noise.

A decoder for generating an audio output signal having one or more audio output channels is provided. The decoder includes a receiving interface for receiving an audio input signal including a plurality of audio object signals, for receiving loudness information on the audio object signals, and for receiving rendering information indicating whether one or more of the audio object signals shall be amplified or attenuated. Moreover, the decoder includes a signal processor for generating the one or more audio output channels of the audio output signal. The signal processor is configured to determine a loudness compensation value depending on the loudness information and depending on the rendering information. Furthermore, the signal processor is configured to generate the one or more audio output channels of the audio output signal from the audio input signal depending on the rendering information and depending on the loudness compensation value. Moreover, an encoder is provided.

The invention provides a volume-based personalized loudness compensation method and an earphone equalization system. The invention provides a method including: by playing a burst sound reference signal and a burst sound test signal at each selected volume, adjusting the gain of each burst sound test signal to make its loudness equal to that of the burst sound reference signal, obtaining a loudnessgain curve at each volume, and correcting and interpolating the loudness gain curve by using a universal iso-loudness curve to form a gain continuous amplitude response curve; and generating the earphone equalization curve by amplitude response curve to compensate the loudness of the audio signal. By using the method and the earphone equalization system provided by the invention, the earphone loudness equalization curve under different volume can be obtained, and the equalization curve is obtained by adjusting the loudness gain of the signal by the user, so as to compensate for the problem ofinsensitive frequency band hearing feeling generated by playback audio under different sound pressure levels, and provide better earphone auditory experience for the user.

An audio processor for processing an audio signal includes: an audio signal modifier for modifying the audio signal in response to a user input; a loudness controller for determining a loudness compensation gain based on a reference loudness or a reference gain and a modified loudness or a modified gain, where the modified loudness or the modified gain depends on the user input; and a loudness manipulator for manipulating a loudness of a signal using the loudness compensation gain.

The present application relates to a sound field expansion method, computer equipment, and a computer-readable storage medium. The method includes the following steps: acquiring at least two channel signals, wherein the at least two channel signals include a first channel signal and a first channel signal. second channel signal; acquiring a first gain signal and a third gain signal corresponding to the first channel signal, and a second gain signal corresponding to the second channel signal; according to the first gain signal and the obtained obtaining a first loudness compensation signal through the second gain signal; obtaining an output signal of the first channel signal according to the third gain signal and the first loudness compensation signal. Loudness compensation is used to remove strong sounds, and to amplify and enhance weaker sounds that cannot be detected by the human ear due to the masking effect, so that they can be perceived by the human ear, improving the three-dimensional sound effect and restoring the real sound field.

The invention discloses a channel-adaptive digital hearing aid wide dynamic range compression method. First, an asymmetric filter bank decomposition and synthesis algorithm is selected by simulating the hearing characteristics of the human ear, and then the filter is designed according to the patient's audiogram and a psychoacoustic model. The number of channels and parameters of the filter group can be used to obtain a personalized filter bank that meets the hearing loss of the patient; finally, the digital hearing aid using the above method performs wide dynamic range compression on the input sound signal sub-channels; the specific steps of compression include: automatically compressing the input signal The adaptive channel filter bank is decomposed to obtain the signal of the adaptive channel; the loudness compensation is performed on each channel signal after decomposition; the filter bank synthesis is performed on the compensated channel signal to obtain a full-band signal; the integrated full-band signal is The signal is converted to an audio signal output. While meeting the performance requirements, the computational complexity of the system is reduced, and the patient's speech intelligibility is improved.