33 results about "Audio normalization" patented technology

In an encoder, multiple channels of audio information representing multidimensional sound fields are split into subband signals and the subband signals in one or more subbands are combined to form composite signals. The composite signals, the subband signals not combined into a composite signal and information describing the spectral levels of subband signals combined into composite signals are assembled into an encoded output signal. The spectral level information conveys either the amplitude or power of the combined subband signals or the apparent direction of the sound field represented by the combined subband signals. In digital implementations, adaptive bit allocation may be used to reduce the informational requirements of the encoded signal.

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.

Several audio signal processing techniques may be used in various combinations to improve the quality of audio represented by an information stream formed by splice editing two or more other information streams. The techniques are particularly useful in applications that bundle audio information with video information. In one technique, gain-control words conveyed with the audio information stream are used to interpolate playback sound levels across a splice. In another technique, special filterbanks or forms of TDAC transforms are used to suppress aliasing artifacts on either side of a splice. In yet another technique, special filterbanks or crossfade window functions are used to optimize the attenuation of spectral splatter created at a splice. In a further technique, audio sample rates are converted according to frame lengths and rates to allow audio information to be bundled with, for example, video information. In yet a further technique, audio blocks are dynamically aligned so that proper synchronization can be maintained across a splice. An example for 48 kHz audio with NTSC video is discussed.

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.

The invention discloses an audio information processing method which is used for solving the technical problem that in the prior art, the display effect of an electronic device is poor. The method includes the steps that when a voice file is output, and M segments of audio information with a first vocal print characteristic in the voice file are analyzed; the M segments of audio information are compared with N segments of audio samples, first audio samples corresponding to the vocal print characteristic same as the first vocal print characteristic in the N segments of audio samples are determined, and first user identification information corresponding to the M segments of the audio information is determined according to the correspondence relation between the audio samples and the user identification information; the voice file is output; when the audio information with the first vocal print characteristic is played, the first display effect of an electronic device is controlled to display the first user identification information. The invention further discloses the electronic device used for achieving the method.

The invention discloses a method and apparatus for adjusting the output audio of a sound box. The method for adjusting the output audio of the sound box comprises the following steps: collecting current position information of a user, determining a current audio parameter corresponding to the current position information, and sending an adjustment instruction of the audio parameter to the sound box, so that the sound box outputs audio whose audio parameter is the current audio parameter according to the adjustment instruction. An audio playing device collects the current position information of the user before controlling the sound box to output the audio information, determines the current audio parameter of the to-be-output audio according to the current position information of the user, sends the adjustment instruction to the sound box, so that the sound box outputs audio information in which the audio parameter is the determined current audio parameter; and the scheme provided by the invention can provide audio information with excellent sound quality effect for the user according to the position information of the user, thereby enhancing the auditory experience of the user on the audio playing device and improving the user experience.

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 discloses an audio content correction method and an audio content correction intelligent device. The audio content correction method comprises the steps of: acquiring audio information sung by a user, performing fundamental frequency analysis on the audio information to obtain fundamental frequency features, and performing rhythm analysis on the audio information to obtain rhythm features, comparing the rhythm features with a template to obtain a time offset sequence, comparing the fundamental frequency features with a template according to the time offset sequence so as to obtain a pitch difference sequence, and performing tone modulation and speed change processing on the audio information of the user according to the time offset sequence and the pitch difference sequence to obtain a corrected audio. By correcting the rhythm and pitch separately, the technical problem that the rhythm and pitch on a timeline interfere with each other and cannot be accurately corrected inoverall analysis in the related art is solved.

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 apparatus for adapting input audio information, encoding one or more audio objects, to obtain adapted audio information is provided. The input audio information includes two or more input audio downmix channels and further includes input parametric side information. The adapted audio information includes one or more adapted audio downmix channels and further includes adapted parametric side information. The apparatus includes a downmix signal modifier for adapting, depending on adaptation information, the two or more input audio downmix channels to obtain the one or more adapted audio downmix channels. Moreover, the apparatus includes a parametric side information adapter for adapting, depending on the adaptation information, the input parametric side information to obtain the adapted parametric side information.

The invention relates to an audio information processing method and a conference terminal thereof. The audio information processing method comprises the steps of: picking up a sound; determining a microphone closest to a sound source according to the sound, determining a sound source distance according to audio information obtained by the microphone and a prestored corresponding relation between a sound pressure level difference value and a distance ratio, and outputting the audio information and distance information; and performing power amplification processing on the audio information according to the distance information. Since the distance of the sound source is judged according to the difference between sound pressure values picked up by different microphones, and the audio information is subjected to amplification processing by means of a power amplification module according to the sound source distance, the conference terminal can automatically adjust volume whether the sound source is far away from or close to the conference terminal, and the volume of the output sound is relatively stable.

The invention discloses a method and a system for audio optimization. The method comprises the following steps of selecting a microphone pickup and picking up a sound analog signal; filtering noise except sound in the sound analog signal and amplifying the sound analog signal; sampling the sound analog signal after noise reduction and amplification processing, and converting the sound analog signal into a data signal; carrying out frequency mixing on the data signal, setting functions and parameters, and carrying out frequency adjustment and frequency conversion mixing; converting a processed digital signal into the analog signal; adding the processed analog signal on a carrier signal and forming a mixed carrier signal after amplification; and emitting the modulated mixed carrier signal through an antenna circuit. According to the method and the system, through changing the input audio signal, a real-time voice change effect is realized, and the synchronous optimization processing of the sound is carried out.

A method of adjusting a television program volume and an apparatus thereof, and a digital television receiving terminal are disclosed. In an embodiment of the invention, through detecting a level amplitude of a peak value volume of an audio signal to be output of a current channel, according to audio information corresponding to the peak value volume, a reference level amplitude of a reference channel corresponding to the audio information is acquired and a specific value of the reference level amplitude and the level amplitude of the peak value volume is acquired too. According to the specific value, a volume of the audio signal to be output is adjusted so that the volume of the audio signal to be output corresponding to each channel of each audio information is consistent with a volume corresponding to the audio information of the reference channel. Volumes output by the channels of the digital television receiving terminal are consistent so that audience hearing discomfort caused by a too loud volume or a too small volume output by the different channels of the digital television receiving terminal is not generated.

The invention discloses a digit watermark based echo inhibition method and system. The method comprises that first audio information is calculated according to obtained far-end audios and pre-generated watermark information, and played via an audio play unit; an audio collection unit collects audios to obtain second audio information; a characteristic sequence is extracted from the second audio information; the correlation degree between temperature audio watermark information and the characteristic sequence is calculated; whether the second audio information includes an echo is determined according to the correlation degree; and if the second audio information is determined to be the echo, the second audio information is zeroed. Echo inhibition is carried out on the watermark information on the basis of frequency domain and discrete logarithm transformation, the audio quality is not decreased, high robustness is provided for audio loss, a watermark can be detected normally under relatively low embedding intensity, and the echo can be detected more accurately.

The present invention provides a smart home control method based on audio signal detection. The method comprises the steps of: establishing connection between at least one home smart device and an audio signal detection server, wherein the audio signal detection server receives audio signals emitted to the home smart device by users and identifies whether the audio signals is an audio signal control command or not; if yes, converting the audio signal control command to control signals which can be identified by the home smart device, and transmitting the control signals to the home smart device, wherein the home smart device receives control signals and executes corresponding control operation. The method provided by the invention detects whether the received audio signals are audio information or not and identifies whether the received audio signals are a control command or not so as to improve the audio identification accuracy and provide convenience for users' usage.

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.

A method and an apparatus for providing digital quality transmission of audio and non-audio information using low cost components and arrangement. The present invention provides for the transmission of the audio and non-audio transmission by first converting the data to conform to the CD standard format and conditioning the converted signal to thereby generate a conditioned EFM signal. The conditioned EFM signal is used to frequency modulate a carrier. By converting the audio and non-audio information to conform to the CD standard format, the present method provides a low cost means of transmitting the data with error detection and correction. Another aspect of the present invention relates to embedding the non-audio information in the SUBCODE block of the data frame according to the CD standard. The non-audio information may be unrelated to the audio information, and the audio and non-audio information may be transmitted to separate devices. Another aspect of the present invention relates to controlling a voltage controlled oscillator in response to the fill level of a file buffer in the decoder.

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 discloses a system and a method configured to analyze audio information. The system and method may include determining for an audio signal, an estimated pitch of a sound represented in the audio signal, an estimated chirp rate (or fractional chirp rate) of a sound represented in the audio signal, and / or other parameters of sound(s) represented in the audio signal. The one or more parameters may be determined through analysis of transformed audio information derived from the audio signal {e.g., through Fourier Transform, Fast Fourier Transform, Short Time Fourier Transform, Spectral Motion Transform, and / or other transforms). Statistical analysis may be implemented to determine metrics related to the likelihood that a sound represented in the audio signal has a pitch and / or chirp rate (or fractional chirp rate). Such metrics may be implemented to determine an estimated pitch and / or fractional chirp rate.

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.

An audio processing method includes the following steps: receiving audio information; capturing a square wave signal, a white signal, and a speech signal of the audio information; calculating a loudness value of the audio information; calculating a first sound quality value and a second sound quality value of the audio information by using the square wave signal, the white signal, and the speech signal of the audio information; calculating a sound quality level of the audio information by using a first calculation formula, the first calculation formula being the loudness value*[1+(B*the first sound quality value+C*the second sound quality value)], where B and C are respectively values greater than 0 and less than 0.1; and displaying a value of the sound quality level of the audio information.

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 a method and device for displaying audio information, and belongs to the field of communication. The method comprises the steps of generating a sine curve model in advance; acquiring the audio information in real time, and resolving the audio information to obtain a frequency; converting the frequency to amplitude of the sine curve model according to a preset corresponding relation; and displaying the sine curve model, and adjusting the amplitude of the sine curve model to the amplitude obtained by conversion. The device comprises a generation module, an acquisition module, a conversion module and a control module. By the method and the device, the change of an audio signal can be displayed in an image, and is more clear and intuitive, the limitation of a single and boring traditional display mode is overcome, the display effect of the audio information is improved, and a user experience is greatly enhanced.

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.

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.