356results 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.

An encoder comprises an input for inputting frames of an audio signal in a frequency band, an analysis filter dividing the frequency band into lower and higher frequency bands, a first encoding block for encoding the audio signals of the lower frequency band, a second encoding block for encoding the audio signals of the higher frequency band, and a mode selector for selecting an operating mode for the encoder among at least a first mode where signals only on the lower frequency band are encoded, and a second mode where signals on both the lower and higher frequency band are encoded. The encoder has a scaler to gradually change the encoding properties of the second encoding block in connection with a change in the operating mode of the encoder. The invention also relates to a device, a decoder, a method, a module, a computer program product, and a signal.

One or more embodiments of the disclosure provide systems and methods for providing media presentations to users of a media presentation system. A media presentation generally includes a one or more media streams provided by one or more capturing users of the media presentation system. In one or more embodiments, a user of the media presentation system may share a media presentation with a viewing user. The media presentation system can provide a number of features to produce a media stream within a media presentation.

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.

The invention provides methods and devices for stereo encoding and decoding using complex prediction in the frequency domain. In one embodiment, a decoding method, for obtaining an output stereo signal from an input stereo signal encoded by complex prediction coding and comprising first frequency-domain representations of two input channels, comprises the upmixing steps of: (i) computing a second frequency-domain representation of a first input channel; and (ii) computing an output channel on the basis of the first and second frequency-domain representations of the first input channel, the first frequency-domain representation of the second input channel and a complex prediction coefficient. The upmixing can be suspended responsive to control data.

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).

To form an audio signal, frequency components of the audio signal which are allotted to a first subband are formed by means of a subband decoder using supplied fundamental period values which respectively indicate a fundamental period for the audio signal. Frequency components of the audio signal which are allotted to a second subband are formed by exciting an audio synthesis filter using an excitation signal which is specific to the second subband. To produce this excitation signal, an excitation signal generator derives a fundamental period parameter from the fundamental period values. The fundamental period parameter is used by the excitation signal generator to form pulses with a pulse shape which is dependent on the fundamental period parameter at an interval of time which is determined by the fundamental period parameter and to mix them with a noise signal.

The invention relates to a method of synthesizing a mono audio signal 3 based on an available encoded multichannel audio signal 2. The encoded multichannel audio signal 2 is assumed to comprise at least for a part of an audio frequency band separate parameter values for each channel of the multichannel audio signal. In order to reduce the processing load in synthesizing the mono audio signal 2, it is proposed that the parameter values of the multiple channels are combined at least for a part of an audio frequency band in the parameter domain. The combined parameter values are then used for synthesizing the mono audio signal. The invention relates equally to a corresponding audio decoder, to a corresponding coding system and to a corresponding software program product.

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.

An electroacoustic transducer includes a magnetic circuit; a frame enclosing the magnetic circuit; and a diaphragm including a center vibrating portion and an outer circumferential vibrating portion, the diaphragm comprising: a first plate thickness area including an entirety of the center vibrating portion, having a double-layer structure of a wood sheet and a reinforcement sheet; and a second plate thickness area including the edge portion of the outer circumferential vibrating portion, having a plate thickness thinner than a thickness of the first plate thickness area, and having a single-layer structure of the reinforcement sheet.

One or more embodiments of the disclosure provide systems and methods for providing media presentations to users of a media presentation system. A media presentation generally includes a one or more media streams provided by one or more capturing users of the media presentation system. In one or more embodiments, a user of the media presentation system may share a media presentation with a viewing user. The media presentation system can provide a number of features to produce a media stream within a media presentation.

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.

The application relates to HFR (High Frequency Reconstruction / Regeneration) of audio signals. In particular, the application relates to a method and system for performing HFR of audio signals having large variations in energy level across the low frequency range which is used to reconstruct the high frequencies of the audio signal. A system configured to generate a plurality of high frequency subband signals covering a high frequency interval from a plurality of low frequency subband signals is described. The system comprises means for receiving the plurality of low frequency subband signals; means for receiving a set of target energies, each target energy covering a different target interval within the high frequency interval and being indicative of the desired energy of one or more high frequency subband signals lying within the target interval; means for generating the plurality of high frequency subband signals from the plurality of low frequency subband signals and from a plurality of spectral gain coefficients associated with the plurality of low frequency subband signals, respectively; and means for adjusting the energy of the plurality of high frequency subband signals using the set of target energies.

Provided is a residual signal coding / decoding apparatus and method. The residual signal coding apparatus includes a transformer, a band splitter, a pulse searcher, and a pulse quantizer. The transformer transforms time-domain residual signals into a frequency domain to output transform coefficients. The band splitter splits the transform coefficients into bands to output the transform coefficients. The pulse searcher searches the transform coefficients for the respective bands to select optimal pulses and output parameters of the optimal pulses. The pulse quantizer quantizes the parameters of the optimal pulses.

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.