33 results about "Audio normalization" patented technology

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.

According to one embodiment, an audio output apparatus, includes: a receiver configured to receiver audio information from an audio processor; an output module configured to output the received audio information; a cable configured to connect between the output module and the receiver to communicate the audio information; a tension detector configured to detect whether tension of greater than or equal to a threshold value is applied to the cable; a determination module configured to determine, when the tension detector detects tension of greater than or equal to the threshold value, a control command to be executed by the audio processor; and a transmitter configured to transmit the determined control command to the audio processor.

The present application provides an audio information processing method and apparatus. The method includes: determining a first camera; acquiring first audio information collected by the first audio collecting unit; acquiring second audio information collected by the second audio collecting unit; processing the first audio information and the second audio information to obtain third audio information, where for the third audio information, a gain of a sound signal coming from a shooting direction of the first camera is a first gain, and for the third audio information, a gain of a sound signal coming from an opposite direction of the shooting direction is a second gain, and the first gain is greater than the second gain; and outputting the third audio information. When the method or the apparatus of the present application is adopted, in synchronously output audio information, volume of a target sound source in a final video image is higher than volume of noise or an interfering sound source outside the video image.

A computer audio system includes an audio codec and a tone controller. The audio codec is operably coupled to receive audio information, which includes tone control settings, PCM digital audio inputs and PCM digital audio outputs. In addition, the audio codec may receive audio information as analog input signals via a line-in, a CD input, or an auxiliary input. Based on the audio information, the audio codec provides a first stereo output, a second stereo output and a monotone audio output. The tone controller is operably coupled to the audio codec and includes a low pass filter, a high pass filter, a band pass filter, and a summing module. The low pass filter is operably coupled to filter the monotone audio output and isolates bass components of the audio signal being processed. By further coupling a volume control module to the low pass filter, the bass component of the audio signal being processed may be varied. The high pass filter is operably coupled to filter the first stereo audio signal to pass treble components of the audio signal being processed. Similarly, a volume control module may be coupled to the high pass filter to provide tone control for the treble components of the audio signal being processed. The band pass filter is operably coupled to filter the second stereo audio output, which passes midband components of the audio signal being processed. Similarly, a volume control module may be coupled to the band pass filter such that midband components of the audio signal being processed may be adjusted. The summing module sums the bass component, treble component and midband component of the audio signal being processed to produce a tone controlled audio output.

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.

A method for generating an audio output from an audio amplifier, the method consisting of receiving a segment of an input audio data stream into a buffer, identifying an adjustment interval in the segment, and calculating an average energy of at least a section of the audio data in the buffer subsequent to the adjustment interval in the segment. The method further includes determining a constant amplification factor in response to the average energy and to a pre-set volume level of the audio output, outputting the audio data from the buffer to the audio amplifier, and, when the audio data output to the audio amplifier reaches the adjustment interval, adjusting the audio amplifier to apply the amplification factor to the audio data in at least the section subsequent to the adjustment interval.

The invention relates to an audio processing system and an audio processing method. The audio processing system is applied to electronic equipment which is used for receiving audio information. The audio information includes a first signal and a second signal, the first signal is a signal affecting the audio output effect of the electronic equipment, and the amplitude of the second signal is greater than that of the first signal. The electronic equipment includes a processor and a memory. The memory stores a preset zero crossing rate, a first amplitude and a second amplitude, all of which represent features of the first signal, wherein the first amplitude and the second amplitude respectively represent the maximum amplitude and the minimum amplitude of the first signal. The audio processing system includes an acquisition module used for acquiring audio information, a dividing module used for dividing the audio information into a plurality of audio segments, a reading module used for reading the zero crossing rates and the amplitudes of speech signals in the audio segments, a judging module used for judging whether the speech signal in the current audio segment is the first signal, and a processing module used for suppressing the first signal to eliminate the first signal.

The invention provides an ultra-shortwave direction-finding station flight correction data acquisition and processing method. The method comprises the steps that (1) before data information acquiring, video, audio and differential GPS timing are carried out; (2) direction-finding station audio and video information and airborne differential GPS information are acquired; (3) the audio information is analyzed and processed to form a data sequence under a time identity, wherein the data sequence comprises communication frequency and communication duration; (4) video information image analysis, segmentation, contrast and recognition are carried out to form an azimuth data set under the time identity; (5) azimuth error processing under specific height and distance after aligned screening is carried out on a video after timing and the data set after GPS resolving according to a step audio data sequence, and error comparative analysis is carried out on the calculated data and direction-finding station correction technical parameter requirement in military standards; and (6) device calibration commissioning is carried out according to a error value for disqualification, and then flight correction verifying is carried out. According to the processing method provided by the invention, the directional precision of a device can be improved; the number of times of flight correction is reduced; in a data acquisition process, the operation state of the device is not affected; and a solid technical support is provided for flight.

The invention relates to a remote interaction method and system. The remote interaction method comprises first-level audio collection, second-level audio collection and third-level audio collection. The first-level audio collection comprises collection of first audio information and collection of second audio information; the second-level audio collection comprises the step of collecting audio information collected by the first-level audio collection, and the audio information is local audio information; the third-level audio collection comprises the step of collecting audio information sent by a far end; the first audio information and the second audio information acquired by the secondary audio acquisition are not played locally; the third-level audio collects the collected audio frequency information to carry out local audio frequency playing; the second-level audio collection performs mutual exclusion collection on the first audio information and the second audio information at thesame time; and after echo cancellation is carried out on part or all of the first-level audio information collected by the second-level audio collection, the first-level audio information is sent toa far end. Compared with the prior art, the audio quality in the remote audio interaction process can be effectively ensured, and a better audio playing effect is achieved.

The invention discloses an audio indexing method based on a multi-distance sound sensor. In the method, a multi-distance sound sensor is used as an audio recording device for recording the audio information in a multimedia conference, a space multi-delay feature is extracted based on the multi-distance sound sensor as a feature for distinguishing different speakers, and a new flow-type algorithm is adopted to perform dimension reduction of the multi-delay feature and classify the speakers according to the identities. The method can reduce the complexity and calculation cost of the system; finally, the audio segment and identity of each speaker are output by the system as audio index information; the optimal discriminant vector set theory obtained by the method can achieve optimal discrimination theoretically; and the method can be applied to a multi-people multi-party conversion scene in a complicated acoustic environment.

An electronic apparatus includes an array of microphones for detecting audible sounds generated by a patient and for generating audio information representing the detected audible sounds, a first beamformer having a first adaptability speed and configured to generate first audio information and first noise information from the audio information, a second beamformer having a second adaptability speed which is slower than the first adaptability speed, the second adaptive beamformer configured to generate second audio information and second noise information from the audio information, an audio classification unit for generating audio classification information based on the first audio information, a head movement detection unit for generating head movement information based on at least one of the second audio information, the first noise information, and the second noise information, and a diagnosis unit for determining a sleep apnea diagnosis based on the audio classification information and the head movement information.

The invention relates to a processing method and device for video and audio information, and a television. The method comprises the steps that a target object at the current time is detected, and a current position value of the target object at the current time is obtained; weighing is conducted on the current position value, and a current weighing position value of the target object is obtained; the current weighing position value is compared with a reference value, and the reference value comprises a maximum position value, minimum position value and a reference position value; when the current weighing position value is located between the maximum position value and the minimum position value, a corresponding regulatory factor of a video and audio parameter is obtained according to the current weighing position value and the reference position value; and regulatory processing is conducted on the video and audio information according to the regulatory factor of the video and audio parameter. According to the processing method, device and television, automatic processing for the video and audio information of the television is achieved; and the processing complexity of the video and audio information of the television is reduced.

To obtain a digital recorder in which noise appearing in a reproduced sound is suppressed when audio encoding conditions (stereo / monoral) are altered. The digital recorder comprises a video encoding means , an audio mode decision means , an audio encoding means , a system encoding means for multiplexing an encoded video / audio, and a recording control means 105. When audio information is varied, the recording control means varies the audio encode mode after lowering the audio output level gradually and then increases the audio output level gradually again.