352results about How to "Improve audio quality" patented technology

In an improved system for receiving digital voice signals from a data network, a jitter buffer manager monitors packet arrival times, determines a time varying transit delay variation parameter and adaptively controls jitter buffer size in response to the variation parameter. A speed control module responds to a control signal from the jitter buffer manager by modifying the rate of data consumption from the jitter buffer, to compensate for changes in buffer size, preferably in a manner which maintains audio output with acceptable, natural human speech characteristics. Preferably, the manager also calculates average packet delay and controls the speed control module to adaptively align the jitter buffer's center with the average packet delay time.

A method for verifying specific profile, legal, and social preference data responses for generating a selectively approved membership database. Membership security is completed with a biometric security authentication of each member upon member logon. A media player is used for recording and delivering video on demand; live video broadcast communications, and video/audio relational database management as well as searching the database for Digital video and audio content defined by indexed unique hint track data. Live video conferencing is available utilizing bandwidth scheduling methods for quality video control and creating audit logs for member accountability and content control.

An audio signal decoder includes a transform domain path configured to obtain a time-domain representation of a portion of an audio content on the basis of a first set of spectral coefficients, a representation of an aliasing-cancellation stimulus signal and a plurality of linear-prediction-domain parameters. The transform domain path applies a spectrum shaping to the first set of spectral coefficients to obtain a spectrally-shaped version thereof. The transform domain path obtains a time-domain representation of the audio content on the basis of the spectrally-shaped version of the first set of spectral coefficients. The transform domain path includes an aliasing-cancellation stimulus filter to filter the aliasing-cancellation stimulus signal in dependence on at least a subset of the linear-prediction-domain parameters. The transform domain path also includes a combiner configured to combine the time-domain representation of the audio content with an aliasing-cancellation synthesis signal to obtain an aliasing reduced time-domain signal.

A method and apparatus for quantizing audio signals is disclosed which advantageously produces a quantized audio signal which can be encoded within an acceptable range. Advantageously, the quantizer uses a scale factor which is interpolated between a threshold based on the calculated threshold of hearing at a given frequency and the absolute threshold of hearing at the same frequency.

Embodiments of the present invention provide an audio signal coding and decoding method and device. The coding method includes: dividing a frequency band of an audio signal into a plurality of sub-bands, and quantifying a sub-band normalization factor of each sub-band; determining signal bandwidth of bit allocation according to the quantified sub-band normalization factor, or according to the quantified sub-band normalization factor and bit rate information; allocating bits for a sub-band within the determined signal bandwidth; and coding a spectrum coefficient of the audio signal according to the bits allocated for each sub-band. According to embodiments of the present invention, during coding and decoding, signal bandwidth of bit allocation is determined according to the quantified sub-band normalization factor and bit rate information. In this manner, the determined signal bandwidth is effectively coded and decoded by centralizing the bits, and audio quality is improved.

An encoder, based on a combination of two audio channels, obtains a first combination signal as a mid-signal and a residual signal derivable using a predicted side signal derived from the mid signal. The first combination signal and the prediction residual signal are encoded and written into a data stream together with the prediction information. A decoder generates decoded first and second channel signals using the prediction residual signal, the first combination signal and the prediction information. A real-to-imaginary transform may be applied for estimating the imaginary part of the spectrum of the first combination signal. For calculating the prediction signal used in the derivation of the prediction residual signal, the real-valued first combination signal is multiplied by a real portion of the complex prediction information and the estimated imaginary part of the first combination signal is multiplied by an imaginary portion of the complex prediction information.

A technique to enhance audio quality of a quantized audio signal when a perceptual audio coder is operating at low bit rates. The perceptual audio coder uses a modified two-loop quantization technique that maintains audio quality at medium to high bit rates while eliminating artifacts at low bit rates. The perceptual audio coder saves vanishing bands by stealing bits from surviving bands to reduce artifacts at low bit rates.

Throttles electronic devices based on compliance with rules for electronic communications such as emails, texts, or postings on social media sites. Rules may for example prohibit certain topics, language, or behaviors such as online bullying. If system detects a violation of the electronic communications rules, it may block access or reduce performance on one or more electronic devices as a consequence. In some cases, devices or selected apps or services may continue to function, but at a reduced level. Conversely the system may provide rewards for conforming with the rules. Throttling of devices may also depend on other factors, such as homework completion, test results, grades, and environmental conditions. Machine learning techniques may be applied to determine when electronic communications may violate the rules. For example, probabilistic topic models may be applied to determine the topics of electronic communications, and to assess whether these topics violate the rules.

A music information processing means comprises a music editing means for assigning an arbitrary sound pattern selected from a plurality of sound patterns in a sound buffer to at least one track, a sound presentation trial processing means for outputting the sound pattern assigned to the track on a speaker of a display monitor via SPU when a control input of the user satisfies a predetermined condition, and an audio data processing means for registering audio data extracted from audio data introduced from a music CD (Compact Disc) or the like loaded in an optical disk drive as one of the sound patterns in a sound pattern file in the sound buffer.

An encoder for a multi-channel audio signal which comprises a down-mixer (201, 203, 205) for generating a down-mix as a combination of at least a first and second channel signal weighted by respectively a first and second weight with different amplitudes for at least some time-frequency intervals. Furthermore, a circuit (201, 203, 209) generates up-mix parametric data characterizing a relationship between the channel signals as well as characterizing the weights. A circuit generates weight estimates for the encoder weights from the up-mix parametric data; and comprises an up-mixer (407) which recreates the multi channel audio signal by up-mixing the down-mix in response to the up-mix parametric data, the first weight estimate and the second weight estimate. The up-mixing is dependent on the amplitude of at least one of the weight estimate(s).

Controls electronic devices and/or throttles electronic devices and/or information for electronic devices based on agreements or events and/or based on schedule, location, time, or any combination thereof. An agreement includes condition(s) to satisfy the agreement, such as task(s) or activities to be performed by an agreement performer, for example based on a schedule and/or location, at a particular time, or time period, or, or events that may be detected, and actions performed to enforce or assert the agreement. Actions may include controlling the electronic device and/or at least partially enabling/disabling or otherwise limiting, reducing or increasing the amount or type of information allowed with respect to any or all electronic devices associated with the agreement performer. Embodiments may reduce or block information from particular sources during a scheduled school class to limit web surfing for information related to that class based on a schedule and/or location of the electronic device.

An apparatus for generating a data stream having a series of segments using data organized in subsequent data frames. The apparatus comprises a packetiser for packetising data from a data frame into the series of segments having a first segment and a second segment, where the second segment has interpretable data entities and has a data entity fragment, the data entity fragment including only a part of an interpretable data entity preceding an interpretable data entity. The apparatus comprises furthermore an information block adder for adding an information block associated with the second segment, the information block indicating an entry point into the second segment, the entry point indicating a start of the interpretable data entity following the data entity fragment.

An apparatus for generating a data stream having a series of segments using data organized in subsequent data frames, a data frame having more important and less important data. It comprises a packetiser for packetising data from a data frame into the series of segments having a first segment and a second segment, where the packetiser is operative to packetise the data of the frame so that a starting point of the more important data coincides with a starting point of the first segment and an information block adder for adding a first information block to the first segment.

A universal reconfigurable system and method are provided for reducing nonlinear echo and residual echo, and for echo leakage prevention in various time-varying and complex environments is proposed in this invention. According to one embodiment an echo cancellation system includes (1) an adaptive linear filter implemented in either time-domain or frequency-domain; (2) a nonlinear echo suppression; (3) echo leakage prevention; (4) direct current (DC), low frequency residual echo and noise reduction; (5) time-domain nonlinear processor (NLP) to reduce the residual echo; and (6) frequency-domain NLP to further reduce the residual echo. The echo cancellation system is universally applicable to acoustic echo cancellation (AEC) and/or electrical echo cancellation applications. In terms of AEC, this invention is reconfigurable for one or more microphones and/or one or more reference channels. The numbers of microphones and reference channels are user configurable.

A perceptual encoder divides an audio signal into successive time blocks, each time block is divided into frequency bands, and a scale factor is assigned to each of ones of the frequency bands. Bits per block increase with scale factor values and band-to-band variations in scale factor values. A preliminary scale factor for each of ones of the frequency bands is determined, and the scale factors for the each of ones of the frequency bands is optimized, the optimizing including increasing the scale factor to a value greater than the preliminary scale factor value for one or more of the frequency bands such that the increase in bit cost of the increasing is the same or less than the reduction in bit cost resulting from the decrease in band-to-band variations in scale factor values resulting from increasing the scale factor for one or more of the frequency bands.

The invention relates to wireless communication between listening devices, in particular to a hearing aid system comprising a hearing instrument and a telephone apparatus. The invention further relates to a method of receiving a telephone call and to the use of a hearing aid system. The object of the present invention is to provide a relatively simple, reliable scheme for receiving a telephone call in a hearing instrument. The problem is solved in that the telephone apparatus comprises a transmitter for establishing a short range digital, low-power link to a hearing instrument comprising a corresponding receiver. This has the advantage of providing a relatively simple and low power solution, where at least the incoming part of a telephone conversation is wirelessly transmitted via a low-power link. The invention may e.g. be used for the communication between a hearing instrument and a telephone, where a standard wireless interface for local connectivity, such as a Bluetooth interface, is not technically feasible or not available for other reasons.

A method for dynamically controlling audio signal gain in a communication device starts with determining a level of an input audio signal. The level is then mapped against a table of predetermined corresponding gain targets to determine an appropriate gain target for that level. The gain targets are then applied to the audio signal, which is then output. The invention is particularly suited to speakerphone operation in a communication device, wherein the gain targets provide compression, expansion or limiting so that distant speakers can be heard clearly.

A flexible electret transducer assembly including an electrical backplate and a membrane made of an electret material is disclosed. A plurality of spacers is formed on a surface of the electrical backplate in a longitudinal or latitudinal direction, and the spacers are used for supporting a vibrating room of the membrane. A working area of the membrane is formed between adjacent spacers, and in each of the working area, the space between the electrical backplate and the membrane is smaller than that in a conventional electrostatic speaker. The spacers between the electrical backplate and the membrane are mass produced through a stamping process. Thereby, an accurate space between the electrical backplate and the membrane can be maintained and accordingly the audio quality can be improved. In addition, a speaker including the flexible electret transducer assembly and a method for fabricating the flexible electret transducer assembly are also disclosed.

A button key structure integrated with a speaker consists of a direction key, a lower lid and a speaker. The direction key is hollow and includes an outer key and an inner key. The outer key is slightly smaller than the key opening and is extended outside the casing. The inner key is slightly larger than the key opening and is confined between the casing and a circuit board. The inner key has contact points located at the bottom to correspond to touch switches located on the circuit board. The lower lid couples with the direction key to form a housing compartment to house the speaker. The speaker connects electrically to the circuit board through a conductive rubber.