1287 results about "Digital audio signals" patented technology

Apparatus is described for digitally encoding an input audio signal for storage or transmission. A distinguishing parameter is measure from the input signal. It is determined from the measured distinguishing parameter whether the input signal contains an audio signal of a first type or a second type. First and second coders are provided for digitally encoding the input signal using first and second coding methods respectively and a switching arrangement directs, at any particular time, the generation of an output signal by encoding the input signal using either the first or second coders according to whether the input signal contains an audio signal of the first type or the second type at that time. A method for adaptively switching between transform audio coder and CELP coder, is presented. In a preferred embodiment, the method makes use of the superior performance of CELP coders for speech signal coding, while enjoying the benefits of transform coder for other audio signals. The combined coder is designed to handle both speech and music and achieve an improved quality.

Method and apparatus for identifying broadcast digital audio signals include structure and / or function whereby the digital audio signal is provided to processing structure which is configured to (i) identify a program-identifying code in the received digital audio signal, (ii) identify a program-identifying code in a decompressed received digital audio signal, (iii) identify a feature signature in the received digital audio signal, and (iv) identify a feature signature in the decompressed received digital audio signal. Preferably, such processing structure is disposed in a dwelling or a monitoring site in an audience measurement system, such as the Nielsen TV ratings system.

This invention relates to a wireless digital transmission system for loudspeakers comprising:compression means for the file representing the digital audio signal of the “compact disc” type, a transmission device comprising means of converting this compressed signal into a series signal moving by packets going to a modulator circuit with phase quadrature and means of transmitting the signals exiting the modulator circuit with phase quadrature to the domestic network for feeding electricity;a receiving device comprising means of connecting to this domestic network and of extracting from the feed electrical signal, by a demodulator with phase quadrature, data packets moving the digital audio signal to convert it into a parallelized digital signal sent to a decompression circuit;means of converting the decompressed digital signals into an analog signal intended to feed a loudspeaker after adequate amplification.

A video, audio and graphics system uses multiple transport processors to receive in-band and out-of-band MPEG Transport streams, to perform PID and section filtering as well as DVB and DES decryption and to de-multiplex them. The system processes the PES into digital audio, MPEG video and message data. The system is capable of decoding multiple MPEG SLICEs concurrently. Graphics windows are blended in parallel, and blended with video using alpha blending. During graphics processing, a single-port SRAM is used equivalently as a dual-port SRAM. The video may include both analog video, e.g., NTSC / PAL / SECAM / S-video, and digital video, e.g., MPEG-2 video in SDTV or HDTV format. The system has a reduced memory mode in which video images are reduced in half in horizontal direction only during decoding. The system is capable of receiving and processing digital audio signals such as MPEG Layer 1 and Layer 2 audio and Dolby AC-3 audio, as well as PCM audio signals. The system includes a memory controller. The system includes a system bridge controller to interface a CPU with devices internal to the system as well as peripheral devices including PCI devices and I / O devices such as RAM, ROM and flash memory devices. The system is capable of displaying video and graphics in both the high definition (HD) mode and the standard definition (SD) mode. The system may output an HDTV video while converting the HDTV video and providing as another output having an SDTV format or another HDTV format.

Techniques like Huffman coding can be used to represent digital audio signal components more efficiently using non-uniform length symbols than can be represented by other coding techniques using uniform length symbols Unfortunately, the coding efficiency that can be achieved by Huffman coding depends on the probability density function of the information to be coded and the Huffman coding process itself requires considerable processing and memory resources. A coding process that uses gain-adaptive quantization according to the present invention can realize the advantage of using non-uniform length symbols while overcoming the shortcomings of Huffman coding. In gain-adaptive quantization, the magnitudes of signal components to be encoded are compared to one or more thresholds and placed into classes according to the results of the comparison. The magnitudes of the components placed into one of the classes are modified according to a gain factor that is related to the threshold used to classify the components. Preferably, the gain factor may be expressed as a function of only the threshold value. Gain-adaptive quantization may be used to encode frequency subband signals in split-band audio coding systems. Additional features including cascaded gain-adaptive quantization, intra-frame coding, split-interval and non-overloading quantizers are disclosed.

With audio data reduction on the basis of ISO / IEC standard 11172-3, a frame length varying by 8 bits is used at a sampling frequency of 44.1 kHz in order to arrive, on average, at a particular fixed data rate. The lengthening of a data frame is signalled by a padding bit in the header of the frames. The invention dispenses with evaluation of the padding bit. Instead, the mean frame length L is calculated, L is rounded down to the next integer, for the subsequent frame it is first established whether the expected sync word for this frame appears, and, if this is so, this frame is decoded without taking into account the padding bit, but if the expected sync word for this frame does not appear, the decoding of the frame is started one 8-bit later without taking into account the padding bit.

A device for prepaid recording of digital audio signals. In a system where digital radio signals are received in an encoded format, a digital converter that is enabled by a smartcard module is employed to convert encoded digital signals to decoded digital signals for recording onto a digital storage media. A smartcard having a prepaid account balance is employed to authorize and control the recording of digital signals. A buffer is advantageously employed to store a quantity of digital signals prior to recording.

The present invention provides an audio headset (1) comprising a resilient headband (2) adapted to extend partially around and grip a wearer s head, the resilient headband (2) having first and second end portions (4),and first and second earphones (6) attached by pliable members (8) to the first and second end portions (4), respectively, wherein the resilient headband is arranged to conduct an audio signal from an audio signal generator to the earphones. The resilient headband can be worn around the back of the head. The audio headset is particularly suitable for use during sports and may be provided in a waterproof version for use whilst swimming. The audio headset may be used with a removable audio player (110) having an integral USB plug (118). The removable audio player (110) may be adapted to receive digital audio signals and power to recharge an internal battery through the USB plug in a first operating mode, and to output analogue audio signals 17 through the USB plug, to the audio headset, in a second operating mode.

The present invention relates to a multi-channel speaker system and a wiring device thereof. The wiring device for a multi-channel speaker system according to an embodiment of the present invention comprises a serial audio signal circuit and a plurality of audio signal separating circuits. The serial audio signal circuit SAC generates a single serial digital audio signal by synthesizing the plurality of analog signals that are generated by the analog audio signal generating circuit, and outputs the same. The audio signal separating circuit receives the serial digital audio signal and separates a corresponding digital signal, and it converts the separated digital signal to an analog signal. Therefore, the plurality of speakers can be connected to the serial audio signal circuit SAC through a single series wire or at least two parallel wires.

The present invention relates to an audio processing system with function of automatic gain control and method thereof. The method includes steps of: receiving a playing command to play an audio file; reading a genre type of the audio file from a tag of the audio file; reading a genre gain value of the genre type from a gain setting table stored in a storage unit; decoding the audio file and generating digital audio signals; converting the digital audio signals to analog audio signals; and signaling a gain amplifier to amplify the analog audio signals by the genre gain value, thereby sounds corresponding to the analog audio signals amplified is proper to a listener's hearing.

Described is a network system and method of use for distributing audiovisual data. The network comprises two or more audio sinks, each of the audio sinks having a corresponding audio format capability, a first source adapted to generate and transmit audiovisual data, wherein the audiovisual data includes an audio file formatted with a first audio format capability, and the at least one source being further adapted to transmit the audiovisual data to the two or more audio sinks through the network system using both an high definition multimedia interface (HDMI) and internet protocol (IP), and a first transmitter adapted to receive the transmitted audiovisual data from the first source. The first transmitter is further adapted to generate two or more transmission streams of audiovisual data, wherein each of the two or more audiovisual data transmission streams include audio data formatted according to the audio format capability of the audio sink to which it is being transmitted.

In a multi-channel digital wireless audio system with at least one transmit node and at least one receive node, each node can both receive and transmit digital audio signals. Signals sent from a receive node to a transmit node may acknowledge satisfactory signal receipt, or may requesting retransmission of data packets received in a corrupted state. Original and retransmitted signals may be sent in compressed form to enable use of narrow-band digital radios. The system preferably incorporates a dual control channel to enable transmission of meta data. Each system node preferably incorporates a hardware-multithreaded processor adapted to implement various functions such as baseband functions, RF protocol functions, error correction functions, and audio processing functions, with each independent thread being adapted to implement a different functional block.

An adaptive “temporal audio scaler” is provided for automatically stretching and compressing frames of audio signals received across a packet-based network. Prior to stretching or compressing segments of a current frame, the temporal audio scaler first computes a pitch period for each frame for sizing signal templates used for matching operations in stretching and compressing segments. Further, the temporal audio scaler also determines the type or types of segments comprising each frame. These segment types include “voiced” segments, “unvoiced” segments, and “mixed” segments which include both voiced and unvoiced portions. The stretching or compression methods applied to segments of each frame are then dependent upon the type of segments comprising each frame. Further, the amount of stretching and compression applied to particular segments is automatically variable for minimizing signal artifacts while still ensuring that an overall target stretching or compression ratio is maintained for each frame.

The present invention provides a system and method for introducing white noises into a digital audio signal so that there is progressive and cumulative degradation in audio quality after each successive reproduction of the audio sound signal in a fashion analogous to analog audio reproduction. The invention provides a white noise generator, and a digital entroping unit. In a preferred embodiment, the white noise generator is implemented by a hardware random number generator. The digital entroping unit controls the magnitude of white noise desired based on a random number generated by the random number generator, and adds the white noise to the input audio sound signal to produce a degraded audio sound signal. The magnitude of white noise can be controlled by using various masking and formatting of random number data.

Distortion modeling produces distortion models for use by a distortion synthesizer to synthesize the harmonic distortion effects of audio distortion devices. A sinusoidal waveform is distorted by an audio distortion device and analyzed using a Fourier transform to produce a distortion model comprising harmonic amplitude and phase parameters. A phase correction process compensates for phase shifts induced by the audio distortion device. The distortion synthesizer uses a distortion function that distorts a digital audio signal according to the distortion model. The distortion model can be modified to alter the distortion effect and can be stored in a data-storage device for later retrieval. A frequency bandsplitter and signal mixer allow the distortion effect to be applied only to the low frequency content of the digital audio signal, thus providing spectral headroom to suppress the production of aliasing noise. Aliasing-noise suppression is provided for a full-bandwidth signal by up-converting the sampling rate of the signal before applying the distortion function and down-converting the sampling rate afterwards. A process is provided to remove the direct-current component that may be induced into the signal by the distortion function.

A video, audio and graphics system uses multiple transport processors to receive in-band and out-of-band MPEG Transport streams, to perform PID and section filtering as well as DVB and DES decryption and to de-multiplex them. The system processes the PES into digital audio, MPEG video and message data. The system is capable of decoding multiple MPEG SLICEs concurrently. Graphics windows are blended in parallel, and blended with video using alpha blending. During graphics processing, a single-port SRAM is used equivalently as a dual-port SRAM. The video may include both analog video, e.g., NTSC / PAL / SECAM / S-video, and digital video, e.g., MPEG-2 video in SDTV or HDTV format. The system has a reduced memory mode in which video images are reduced in half in horizontal direction only during decoding. The system is capable of receiving and processing digital audio signals such as MPEG Layer 1 and Layer 2 audio and Dolby AC-3 audio, as well as PCM audio signals. The system includes a memory controller. The system includes a system bridge controller to interface a CPU with devices internal to the system as well as peripheral devices including PCI devices and I / O devices such as RAM, ROM and flash memory devices. The system is capable of displaying video and graphics in both the high definition (HD) mode and the standard definition (SD) mode. The system may output an HDTV video while converting the HDTV video and providing as another output having an SDTV format or another HDTV format.

A self-descriptive microphone array includes a microphone array memory, such as, for example a ROM, EEPROM, or other conventional memory, which contains a microphone array device description. This device description includes parametric information which defines operational characteristics and configuration of the microphone array. In further embodiments, the microphone array uses any of a variety of conventional wired or wireless computer interfaces, including serial, IEEE 1394, USB, Bluetooth™, etc., to connect to a computing device. Once connected, the microphone array provides its device description to the computing device. Sound processing software residing within the computing device is then automatically configured for optimally interacting with one or more analog or digital audio signals provided by the microphone array. In another embodiment, the microphone array performs integrated self calibration for automatically updating the device description. The self calibration is performed either upon connection to the computing device, or upon regular or user-specified intervals.

A header format information determination section reads header format information to determine whether or not a stream conforms to IEC 61937-6. A header format information conversion section acquires or prepares information on sync words, the sampling frequency, the number of channels and the length of audio data for a stream not conforming to IEC 61937-6, and converts the information to header format information defined in IEC 61937-6. A digital audio signal output section outputs the converted stream.

A wireless transmitter (300) communicates with a wireless earphone (100) to obtain remote communications. The wireless earphone includes: a wireless network card (30′) complying with the 802.11a / b / g standard, for receiving data packets and unpacking them to release digital audio signals; a control processor (20′) connected with the wireless network card, for dealing with the digital audio signals; and a digital-to-analog converter (50) connected with the control processor, for converting the digital audio signals to analog audio signals. The wireless transmitter has a structure similar to that of the wireless earphone, except that the wireless transmitter has an analog-to-digital converter (10) instead of the digital-to-analog converter.

A modular wireless headset includes earpiece(s) and microphone(s), where the earpiece and microphone may be physically separate devices. The earpiece renders inbound radio frequencies received from a host device audible. The earpiece may include a receiver module, data recovery module, and speaker module. The receiver module may convert inbound RF signals into low intermediate frequency (IF) signals. The data recovery module recovers audio signals from the low IF signals. The speaker module renders the audio signals audible. The microphone converts received audio signals into outbound RF signals, where the outbound RF signals are transmitted to the host device. The microphone includes an audio input module and a transmitter module. The audio input module is operably coupled to convert received analog audio signals into digital audio signals. The transmitter module is operably coupled to convert the digital audio signals into the outbound RF signals.

An efficient audio coding device that quantizes and encodes digital audio signals with a reduced amount of computation. A spatial transform unit subjects samples of a given audio signal to a spatial transform, thus obtaining transform coefficients of the signal. With a representative value selected out of the transform coefficients of each subband, a quantization step size calculator estimates quantization noise and calculates, in an approximative way, a quantization step size of each subband from the estimated quantization noise, as well as from a masking power threshold determined from a psycho-acoustic model of the human auditory system. A quantizer then quantizes the transform coefficients, based on the calculated quantization step sizes, thereby producing quantized values of those coefficients. The quantization step sizes are also used by a scalefactor calculator to calculate common and individual scalefactors. A coder encodes at least one of the quantized values, common scalefactor, and individual scalefactors.

A dangle, which plugs into the audio / earpiece jack of host devices, allows them to accept payment cards as a point of sale (POS) terminal. It contains a magnetic stripe (magstripe) card reader, a smart card reader, and / or a proximity card reader (reader(s)); a microprocessor or microcontroller (CPU); and a circuit for drawing power from a digital audio signal (power circuit). Payment card data is collected by the reader(s), passed to the CPU where it is encrypted, encoded, and modulated, sent to the host device through the Microphone input contact of the audio / earpiece jack, then transmitted via one the host device's networks to a merchant account provider for processing. A digital audio signal, constantly generated by the host device, provides power to the dongle via the power circuit. The power circuit converts the digital audio signal to stable DC power with the appropriate voltage for powering the dongle's other components.

A portable assistive listening system for enhancing sound for hearing impaired individuals includes a hearing aid and a separate handheld digital signal processing device. The hearing aid includes a microphone, a digital signal processor, a speaker, a power source, an ultra-wide band (UWB) transceiver, and a control system configured and arranged to monitor a status of the power source and a status of the UWB transceiver, to receive an input signal from the microphone, to selectively route the input signal for processing responsive to the status of the power source and of the UWB transceiver, and to deliver an output signal to the speaker. The handheld digital signal processing apparatus includes a UWB transceiver, and a digital audio signal processor configured and arranged to receive the input signal from the earpiece control system, to process the input signal to enhance the input signal and to output the enhanced signal to the hearing aid control system. The hearing aid and the signal processing apparatus communicate through the UWB transceivers.

In an embodiment, a method of receiving a digital audio signal, using a processor, includes correcting the digital audio signal from lost data. Correcting includes copying frequency domain coefficients of the digital audio signal from a previous frame, adaptively adding random noise coefficients to the copied frequency domain coefficients, and scaling the random noise coefficients and the copied frequency domain coefficients to form recovered frequency domain coefficients. Scaling is controlled with a parameter representing a periodicity or harmonicity of the digital audio signal. A corrected audio signal is produced from the recovered frequency domain coefficients.

The present invention relates to an audio signal controller adapted to receiving first and second digital audio signals and estimating a signal feature of the first or second digital audio signal. The estimated signal feature is compared with a predetermined feature criterion and the audio signal controller switches from conveying the first digital audio signal to conveying the second digital audio signal to a controller output, or vice versa, at a zero-crossing of the first digital audio signal or the second digital audio signal based on the comparison between the estimated signal feature and the predetermined feature criterion.

A system for recording and displaying a multimedia presentation, includes a digital camera having a solid state image sensor for selectively generating a sampled analog video image signal or a higher resolution sampled analog still image signal, and a microphone for generating an analog audio signal. An analog to digital converter converts the sampled analog video image signals and audio signal to a digital image signal and digital audio signal and an audio visual encoder in the camera compresses the digital video signal and associated digital audio signal to form a compressed video bit stream. The camera is operated to periodically capture a higher resolution still image to form a still image file while capturing a video sequence. A pointer linking a captured high resolution still image with a corresponding frame in the compressed video bit stream is appended to the still image file. An object oriented image processing system, includes an image processing computer, an object oriented operating system, an image memory for storing the compressed video bit stream and the still image files as objects, a graphic user interface including a display and operator input device, a decoder for decoding the compressed video bit stream, and an application program for generating low resolution index images from the higher resolution still images with pointers linking the index images to the high resolution still images and storing the index images with their associated pointers as objects in the image memory, for displaying a plurality of the low resolution index images on the graphic user interface and responsive to operator selection of an index image from the displayed index images, employing the pointers stored with the selected index image and the associated high resolution still image, to retrieve a corresponding portion of the compressed motion image, decompressing the retrieved portion of the compressed motion image, and displaying the decompressed portion of the motion image.

Compression of audio signal data is described herein. In various embodiments, the compression of each unit of the audio signal data includes the employment of a distribution substantially representative of a subblock of residual data of the unit of audio signal data, to reduce the amount of data having to be transmitted to transmit the unit of audio signal data to a recipient.

A system and method executed by audio processing software on one or more electronic devices in a computer system to process digital audio signals. The system comprises a digitizer for digitizing a received audio signal; and processor for performing a plurality of audio processing functions on the digitized audio signals, each of the audio processing functions having at least one programmable parameter, and wherein each of the audio processing functions are categorized and grouped as audio objects, and organized into a channel strip, the channel strip processing digitized audio signals for a particular received audio signal, and wherein, the audio objects are fixed in order, so that the digitized received audio signals are processed by a predefined number of N audio objects, and wherein the N audio objects occur in a fixed sequence, and further wherein, the N audio objects comprise a first subset of non-exchangeable audio objects and a second subset of exchangeable audio objects, such that any one or more of the second subset of audio objects can be exchanged by a replacement audio object, and further wherein when the audio processing functions are programmed, they can be saved without compiling the audio processing software.

A digital audio system including a combination of analog and digital volume control is disclosed. A variable power supply voltage biases a power amplifier for each channel, and applies a bias voltage corresponding to an analog volume control signal. In one disclosed embodiment, digital gain control circuitry compares the bias voltage with the level expected for the analog volume control signal; if the bias voltage has not dropped in response to a reduction in the analog volume control signal, the digital gain control circuitry reduces the digital gain of the input digital audio signal, until the bias voltage responds to the reduced volume. In another disclosed embodiment, modeling or characterization of the audio system is used to derive a digital gain control signal based on the desired volume signal and the amplitude of the digital audio signal itself. In another disclosed embodiment, a slew rate limiter slows the rate of change of the control signal applied to the variable power supply, in which case the slowed control signal is used in the comparison or calculations of the reduced digital gain.