Showing papers on "Audio signal processing published in 1995"

Apparatus and methods for including codes in audio signals and decoding

Abstract: Apparatus and methods for including a code (68) having at least one code frequency component in an audio signal (60) are provided. The abilities of various frequency components in the audio signal to mask the code frequency component to human hearing are evaluated (64), and based on these evaluations an amplitude (76) is assigned to the code frequency component. Methods and apparatus for detecting a code in an encoded audio signal are also provided. A code frequency component in the encoded audio signal is detected based on an expected code amplitude or on a noise amplitude within a range of audio frequencies including the frequency of the code component.

Method and apparatus for robust information coding

Abstract: A digital signal is imperceptibly embedded into an input source signal, such as an image or video signal, to produce an encoded (sometimes termed "watermarked") signal. The principle of quasi-rotational symmetry is employed to facilitate detection of the embedded signal notwithstanding rotation of the encoded signal. Single or multiple degrees of symmetry can be employed. In another aspect, the digital signal is transformed to a frequency domain and phase-only filtered prior to its combination with the input source signal. In an illustrative embodiment, this filtering operation helps hide the digital signal within the source signal, and facilitates detection of the embedded digital signal even after the encoded signal has undergone various forms of corruption.

Multi-channel digital transceiver and method

Abstract: A multi-channel digital transceiver (400) receives uplink radio frequency signals and converts these signals to digital intermediate frequency signals. Digital signal processing, including a digital converter module (426), is employed to select digital intermediate frequency signals received at a plurality of antennas (412) and to convert these signals to baseband signals. The baseband signals are processed to recover a communication channel therefrom. Downlink baseband signals are also processed and digital signal processing within the digital converter module (426) up converters and modulates the downlink baseband signals to digital intermediate frequency signals. The digital intermediate frequency signals are converted to analog radio frequency signals, amplified and radiated from transmit antennas (420).

Apparatus and methods for music and lyrics broadcasting

Abstract: A system for broadcasting audio music and broadcasting lyrics for display and highlighting substantially simultaneously with the occurrence of the lyrics in the accompanying audio music is provided. The system includes a audio music source that provides a data output and a analog audio signal output. A computer receives the data output by the music source and generates lyric text data and lyric timing commands. A subcarrier generator generates a subcarrier signal carrying the lyric text data and lyric timing commands. An FM transmitter broadcasts a composite signal that combines the analog output of the music source with the subcarrier signal. A lyric display unit receives the composite signal, separates and decodes the subcarrier signal and displays and highlights lyrics according the lyric text data and lyric timing commands decoded from the subcarrier signal.

Method and apparatus for using visual images to mix sound

Abstract: A method and apparatus for mixing audio signals. Each audio signal is digitized and then transformed into a predefined visual image, which is displayed in a three-dimensional space. Selected audio characteristics of the audio signal, such as frequency, amplitude, time and spatial placement, are correlated to selected visual characteristics of the visual image, such as size, location, texture, density and color. Dynamic changes or adjustment to any one of these parameters causes a corresponding change in the correlated parameter.

The Decorrelation of Audio Signals and Its Impact on Spatial Imagery

Abstract: As used here, the term "decorrelation" refers to a process whereby an audio source signal is transformed into multiple output signals with waveforms that appear different from each other, but which sound the same as the source. In the experience of most sound professionals, decorrelation occurs as a by-product of other acoustic or electronic processes that often change the sound of the source. In acoustic performances, decorrelation occurs as a by-product of reverberation and chorusing, and in digital signal processing, "stereoized" reverberation and chorusing achieve the same effect. Decorrelation occurs in sound synthesis when there are slight differences between the sounds synthesized for the output channels. That often happens with granular synthesis, but can also happen with frequency modulation or additive synthesis if the composer takes special care in designing the algorithms. In the audio industry, there is a long tradition of devices for the home or studio that "stereoize" monophonic signals, and they too typically decorrelate the output channels. Numerous settings on effects processors for flanging, combing, etc. produce decorrelated output. In recording studios, vocal artists sometimes are recorded twice on separate tracks so that the micro-variations in the two performances create decorrelation. Why focus on decorrelation as a separate aspect of these processes? In the field of spatial hearing, signal decorrelation is known to have dramatic impact on the perception of sound imagery. The degree to which sounds are decorrelated has proven to be a significant predictor of perceptual effects, both in natural environments and in audio reproduction. Therefore, all of the diverse processes mentioned above are related to each other by the impact of decorrelation on the spatial imagery of the sound. While there is a considerable literature on spatial sound processing, this literature is usually concerned with one of two goals: (1) positioning sound images at a particular location in three-dimensional space, or (2) creating three-dimensional simulated environments. These goals are important, but there are obviously many other creative potentials for spatial sound processing, and other kinds of practical problems to solve. For example, decorrelation can produce sound images with the width, depth, and spaciousness typical of natural environments while circumventing the computational burden of a full environmental simulation.

Apparatus and methods for including codes in audio signals

Abstract: Apparatus and methods for including a code having at least one code frequency component in an audio signal are provided. The abilities of various frequency components in the audio signal to mask the code frequency component to human hearing are evaluated and based on these evaluations an amplitude is assigned to the code frequency component. Methods and apparatus for detecting a code in an encoded audio signal are also provided. A code frequency component in the encoded audio signal is detected based on an expected code amplitude or on a noise amplitude within a range of audio frequencies including the frequency of the code component.

Digital sub-band transmission system with transmission of an additional signal

Abstract: Reproduction accuracy of a digital signal, representing for example stereo audio signals, is improved by transmitting sample data which have been encoded to form transmission signals. The transmission signals are arranged in consecutive frames, each frame including a plurality of information packets, and each information packet including N bits. The number of information packets in a frame is determined by the ratio of the product of the transmission bit-rate and the number of samples represented in the frame, to the product of the number of bits in a packet and the sampling frequency. The transmission signals are recorded in a record carrier, or are transmitted in real time. In a digital stereo audio signal embodiment, the samples may be sub-band encoded or transform encoded. The transmission signal may include an indicator signal indicating a combination of certain samples, or scale factor signals for specific components of the sampled signal.

Spectral shape analysis in the central auditory system

Abstract: A model of spectral shape analysis in the central auditory system is developed based on neurophysiological mappings in the primary auditory cortex and on results from psychoacoustical experiments in human subjects. The model suggests that the auditory system analyzes an input spectral pattern along three independent dimensions: a logarithmic frequency axis, a local symmetry axis, and a local spectral bandwidth axis. It is shown that this representation is equivalent to performing an affine wavelet transform of the spectral pattern and preserving both the magnitude (a measure of the scale or local bandwidth of the spectrum) and phase (a measure of the local symmetry of the spectrum). Such an analysis is in the spirit of the cepstral analysis commonly used in speech recognition systems, the major difference being that the double Fourier-like transformation that the auditory system employs is carried out in a local fashion. Examples of such a representation for various speech and synthetic signals are discussed, together with its potential significance and applications for speech and audio processing. >

Method and apparatus for audio and video synchronizing in MPEG playback systems

Abstract: A multimedia system includes an integrated system and video decoder with an audio/video synchronization circuit (200) for substantially synchronizing the display of video images with audio playback. An audio presentation time stamp (APTS) (415) is detected in the compressed/coded audio data stream in the integrated system and video decoder and stored in a data latch (442). The compressed/coded audio data stream is fed to an audio decoder which decodes/decompresses the audio data and outputs an audio signal. The audio decoder detects when audio data corresponding to an APTS (415) had been output and sets a corresponding flag (213). The flag (213) indicates to the integrated system and video decoder that a corresponding audio segment had been decoded/decompressed and output. The integrated system and video decoder then synchronizes the video output with the audio output by repeating or skipping frames of video data.

Videophone for simultaneous audio and video communication via a standard telephone line

Abstract: A videophone includes a video camera for transducing and digitizing images at a station and connected to a video processor for compressing and encoding the digitized images into video data and an audio transducer and digitizer connected to an audio processor synchronized with the video processor for compressing and encoding digitized sound. The video and audio processors are connected to a communication controller for formatting and transmitting the video data bundled with the audio data via a standard bidirectional (full duplex) telephone line in the form of a twisted pair. The communication controller can also receive and then unformat and unbundle audio and video data, and the video processor and audio processor can decode and decompress the received video and audio data, and the station further includes a display connected to the video processor for reproducing images and an electroacoustic transducer connected to the audio processor for reproducing sounds transmitted by another station. Rather than format and transmit only delayed or partial changed video data when the image changes significantly, the large amount of changed video data being stored is overwritten when the amount reaches a predetermined level, and new video data corresponding to the present appearance of the image is compressed, encoded, bundled with associated audio data, formatted, and transmitted. The image reproduced at the receiving station therefore jumps ahead to an updated image without intervening blurred images.

Level meter for digitally-encoded audio

Abstract: For professional audio operations such as recording and broadcasting, an LED bar-graph display instrument monitors average and peak loudness levels of stereophonic audio signals that have been encoded in serial digital format such as AES/EBU digital audio format. Internal audio processing circuitry, that can be implemented digitally with a custom chip gate array set, receives as input a serial stream of digital stereo audio data and converts this to ballistically conditioned logarithmic average and peak levels which are simultaneous displayed, generally simulating the ballistics of contemporary standard electronically displayed loudness meters such as the Dorrough analog model. The peak hold can be switched manually or internally between three hold durations: indefinite, 3 seconds or zero. A preferred dual embodiment provides digital implementation driving a pair of LED bar-graph displays side-by-side in vertical or horizontal orientation for stereo applications, and provides selectable display of stereo signals or sum and difference signals. A special peak capture circuit ensures that even very narrow peak levels are indicated at full amplitude despite the controlled ballistic rise rate. Over-range is indicated by a color change of three top display segments.

Audio listen and voice security system

Abstract: A security alarm system including multiple, zone distributed audio monitors and alarm sensors which report and verify detected alarms and communicate with a system controller and central station Ambient audio is continuously and selectively recorded in a storage memory and is replayable to verify alarms detected at the alarm sensors During pre-alarm conditions, all audio inputs are summed and recorded During an alarm state, the audio input of the audio monitor physically closest to a reporting alarm sensor is automatically selected and the post-alarm audio activity is recorded only for that sensor The central station is able to selectively communicate via a phone link with each audio controller and engage in half duplexed voice communication with the alarm site, remotely playback and listen to the locally recorded audio data, or program each audio controller The monitored audio data is stored in RAM storage in discreet segments corresponding to predetermined periods of pre-alarm and post-alarm data

Disposable audio processor for use with implanted hearing devices

Abstract: The echo canceller is designed to be placed between a hands-free acoustical interface and a communications network. It comprises a plurality of processing paths connected in parallel and each allocated to one of a plurality of adjacent sub-bands taken from the spectrum band of the output signal. Each path comprises an analysis filter receiving the echo-containing signal for transmission after correction, a second analysis filter receiving the incoming signal coming from the network, and feeding an adaptive filter that supplies an estimated echo in the respective sub-band to the subtractive input of the subtracter and a synthesis filter. The adaptive filters in at least some of the sub-bands implement a QR decomposition RLS algorithm on the incoming signal, using the fast version thereof, with or without recursive order.

System and method for classification of audio or audio/video signals based on musical content

Abstract: An automated system and method for classifying audio or audio/video signals as music or non-music is provided. A spectrum module receives at least one digitized audio signal from a source and generates representations of the power distribution of the audio signal with respect to frequency and time. A first moment module calculates, for each time instant, a first moment of the distribution representation with respect to frequency and in turn generates a representation of a time series of first moment values. A degree of variation module in turn calculates a measure of degree of variation with respect to time of the values of the time series and produces a representation of the first moment time series variation measuring values. Lastly, a module classifies the representation by detecting patterns of low variation, which correspond to the presence of musical content in the original digitized audio signal, and patterns of high variation, which correspond to the absence of musical content in the original digitized audio signal.

System and method for scaleable streamed audio transmission over a network

Abstract: An audio data transmission system encodes audio files into individual audio data blocks which contain a variable number bits of digital audio data that were sampled at a selectable sample rate. The number of bits of digital data and the input sampling rate are scaleable to produce an encoded bit stream bit rate that is less than or equal to an effective operational bit rate of a recipient's modem. The audio data transmission system uses computing units which are designed to select an appropriate combination of block size and input sampling rate to maximize the available bandwidth of the receiving modem. For example, if the modem connection speed for one modem is 14.4 kbps, a version of the audio data compressed at 13000 bits/s might be sent to the recipient; if the modem connection speed for another modem is 28.8 kbps, a version of the audio data compressed at 24255 bits/s might be sent to the receiver. The audio data blocks are then transmitted at the encoded bit stream bit rate to the intended recipient's modem. The audio data blocks are decoded at the recipient to reconstruct the audio file and immediately play the audio file as it is received. The audio data transmission system can be implemented in online service systems, ITV systems, computer data network systems, and communication systems.

System for tdma mobile-to-mobile vselp codec bypass

Abstract: In a TDMA mobile-to-mobile connection, the end-to-end audio signal quality as well as system performance can be improved by providing digital signal processors the capability to automatically switch configuration such that each digital signal processor in a mobile-to-mobile communication connection can automatically identify a TDMA mobile-to-mobile connection and bypass the speech encoding and decoding processes within the digital signal processors. The two digital signal processors are virtually connected at the channel codecs.

Muting apparatus for a compressed audio/video signal receiver

Abstract: A compressed audio/video, A/V, receiver provides A/V component timing reference signals (208,212,214,220,222) (PTS's) coincident with reproduction of associated decompressed component signals. Synchronization apparatus generates a function of the difference (217-219) of occurring component audio and video PTS's. This function, which is indicative of relative audio and video synchronization, is compared with a predetermined threshold value (225), and a mute control signal is generated when the threshold is exceeded. Muting circuitry (229), responsive to the control signal disables audio reproduction when the reproduced audio and video component signal timing deviate from mutual synchronization.

Hybrid analog/digital STB

Abstract: A hybrid analog/digital STB includes a tuner supplying respective analog and digital television signal processing paths. The analog processing path provides demodulated composite analog video and audio signals and the digital processing path provides demodulated component digital video and audio signals. The demodulated analog and video signals are combined in composite or component form to achieve various desirable effects in a highly flexible system architecture. OSD and display map normalization functions are also integrated within the system architecture.

Apparatus and method for stereo audio encoding of digital audio signal data

Abstract: A stereo audio encoding method for encoding left and right original signals to a left and right reproduced signals for suppressing a loss of quality in the reproduced audio signal. The correlation between the right and left channel signals is determined, and the phase of each signal is compared. If the two signals have the same phase, a modified scale factor is calculated based on a power equalization method, but if the two signals are in opposite phase, another modified scale factor is calculated based on an error minimization method. The modified scale factors are used for calculating the reproduced signals.

Digital, pulse width modulation audio power amplifier with noise and ripple shaping

Abstract: A digital pulse-width modulated method and apparatus is described which is characterized by noise and ripple shaping for purposes of high-fidelity digital power amplification. A sampled power source amplitude signal is used to shape an interpolated input digital signal in a divider, which is then fed to a noise shaper to produce a lower bit digital signal to represent the original input signal. The output power signal is not affected by the requantization error or the power source ripple. Therefore, DC power regulators normally employed in digital pulse-width modulated audio power amplifiers can be entirely eliminated without sacrificing the signal-to-noise ratio.

Audio signal processor providing simulated source distance control

Abstract: An audio signal processing system produces an output 24 having an illusory distance effect for a sound source signal S by feeding it via a direct signal path 25 and an indirect signal path 22, 23 passing through early reflection simulation apparatus 1 which feed an output mixing mechanism 9. A control system adjusts the relative delays 3, 4 and relative gains 5, 6 in the direct 25 and indirect 22, 23 signal paths to modify the illusory distance effect so as to substantially maintain the mathematical relationship between the gains and time delays of simulated reflections relative to first sound arrivals at the output 24 encountered for sounds at that source distance in actual rooms. Signal paths 22, 23, 24, 25 may be stereophonic or multichannel using matrix gain coefficients in the early reflection simulator 1, and may produce different simulated distances for different sound positions. A plurality of sound sources S may have different simulated distances while feeding a common early reflection simulator 1.

Audio reproducing apparatus corresponding to picture

Abstract: Audio reproducing apparatus reproducing an audio signal corresponding to a picture which localizes a sound image in a direction corresponding to the picture by processing an audio signal in a real-time fashion is supplied with an audio signal from a general-purpose signal source such as a laser disc a digital vibratory gyroscope detects a rotational angle of a listener's head. In response to the detected rotational angle, the audio reproducing apparatus subjects the audio signal to a predetermined signal processing in a real-time fashion. Thus, a sound image is localized in the direction corresponding to the picture projected on a screen from a projector.

Audio and video signal distribution system and method

Abstract: A system for distributing one of a plurality of input signals from one location in a building having standard AC power to another location in the building. An input signal is selected by a switcher and, if it is an analog signal, it is converted to a digital signal. The digital signal is then processed to enable it to be superimposed on, and transmitted over, an AC power line to the other location. At the other location the transmitted digital signal is processed again to restore it to its original digital format at which time it can be passed directly to a digital amplifier or converted to an analog signal for further processing.

Method and apparatus for digital mixing of audio signals in multimedia platforms

Abstract: A multimedia method and apparatus is able to digitally mix audio signals to produce combined audio output signals. Prior to digital mixing, the audio input signals are de-formatted using a digital de-formatter, volume adjusted using digital volume controllers and converted to a common sampling rate utilizing a digital interpolator or decimator.

Low disturbance audio for awareness and privacy in media space applications

Abstract: Media spaces are systems designed to use audio, video, and other media to create shared "spaces" in which distributed work groups can operate smoothly and conveniently. This paper presents a new audio processing technique that allows users of a media space to maintain awareness of conversations that they are not directly involved in, without demanding much of their attention, nor violating the privacy of their coworkers. The technique works by processing speech signals into non-speech signals which none the less retain important characteristics of the original speech (such as volume profile and various characteristics of the original speaker such as typical overall frequency distribution). This technique has applications to general awareness in media spaces, as well as to specialized circumstances such as a facility to quickly "stick your head into someone's office" to see if they can be interrupted (without violating their privacy).

Method and apparatus for reading and writing audio and digital data on a magnetic tape

Abstract: A method and apparatus for recording and reading magnetic tape that contains audio signals and data that is related to the audio signals. In the preferred embodiment, the audio signals are music, and the related data are lyrics to the music displayed in real time to the music as it is being played. The lyrics are modulated onto an ultrasonic carrier signal and mixed with the audio signal on one track of the tape. The tape may be played on a conventional tape recorder without having interference from the recorded data at ultrasonic frequencies. In a suitably equipped tape player, the mixed signal is read and filtered to generate a high pass filtered signal and a bandpass filtered signal. The bandpass filtered signal filters out frequencies outside the range of human hearing to generate an audio signal. The high pass filtered signal is demodulated by the carrier frequency to reconstruct the lyrics. A two-track system records an unmodulated carrier frequency on the second track and is subtracted from the modulated carrier frequency to generate the lyrics. This compensates for variations in tape speed that shift the frequency of the data modulated carrier signal.

Software sound synthesis system

Abstract: An audio signal processing system including an input circuit for inputting musical instrument digital interface (MIDI) commands in real time over a plurality of channels, a computer including a central processing unit (CPU) supplied with the MIDI commands for simultaneously synthesizing one or more voices for each of the channels in response to the MIDI commands, each of the voices being generated by one or more of a plurality of predefined audio synthesis algorithms executed in software, a random access memory (RAM) for storing digital voice data representative of each of the voices generated by the CPU, an output circuit for audibly reproducing the voices from the digital voice data stored in the RAM, and wherein the CPU, in generating the voices selects the one or more audio synthesis algorithms based on one or more of the following criteria: the external processing demands placed upon the CPU by other operations being performed by the personal computer, a best match, according to predetermined criteria, between the type of voice required and audio synthesis algorithms available to the CPU, and the availability of wavetable voice data to be buffered into the RAM.

Device and method for automatically controlling sound volume in a communication apparatus

Abstract: A communication apparatus comprises a sound volume controller in which a speech signal is discriminated from background noise of a transmitting audio signal (Sat) and the sound volume control is performed when no speech signal is present in the transmitting audio signal. The sound volume controller comprises an averaging circuit (202, 303) for averaging an amplitude of the transmitting audio signal (Sat), a discriminator (203, 304) for discriminating a speech signal of the transmitting audio signal (Sat) by comparing the transmitting audio signal (Sat) with the averaged amplitude (SL, Sav), and a controller (205, 104) for controlling an amplitude of a receiving audio signal when no speech signal is present in the transmitting audio signal. A level of the averaged amplitude is determined by comparing the averaged amplitude with at least one predetermined reference level, and the amplitude of the receiving audio signal is varied according to the level of the averaged amplitude when no speech signal is present.

Kernel-based digital audio mixer

Abstract: A system and method are disclosed for mixing multiple digital audio streams originating from or destined for multiple tasks in a computer system. In one aspect, a method is carried out in a computer system having a processor. The processor executes a multitasking operating system that has a kernel, executes a first task under control of the operating system, and while executing the first task executes a second task under control of the operating system. First and second audio streams are provided, each made up of a series of digital audio samples representing an audio signal and affiliated, respectively, with the first and second tasks. By executing a portion of the operating system kernel with the processor, the first and second audio streams are mixed to provide a mixed audio stream made up of a series of digital audio samples representing a superposition of the first and second audio signals.