1279 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.

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.

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.

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.

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 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.

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.

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.