Analog Audio Signal Research Articles

Methods and systems for generating phase-derivative sound

Methods and systems for digitally generating sound from phase and amplitude information of a narrow bandwidth signal, such as a narrow bandwidth locator signal. Phase-derivative information is determined from the phase information. The bandwidth of the phase-derivative information is spread out, or stretched, over a wider bandwidth, so that the frequency variations will be more perceptible to users. The result is combined with an audio band carrier frequency, the result of which controls an oscillator. The oscillator output is combined with the amplitude information to generate an analog audio signal that is modulated with the amplitude information and the phase-derivative information. The amplitude information wider bandwidth phase-derivative information are used to modulate an audio carrier in both frequency and amplitude. The overall process can be thought of as a translation of the frequency and amplitude information from the narrow bandwidth around the locate frequency to a wider bandwidth on a chosen carrier frequency in the audio band. The received amplitude and phase information is received at an input sample rate. Where the input sample rate is relatively low, the amplitude and phase information are up-sampled to an output sample rate that is higher than a desired audio frequency. The higher output sample rate insures that there are sufficient samples of the signal during each cycle or period of the audio frequency. The higher sample rate is typically also the sample rate of a digital to analog converter that outputs an analog signal to a speaker. The amplitude information and/or phase derivative information are optionally scaled to system gain. The sound heard by the operator can optionally be adjusted with an optional selectivity filter.

Interface device to couple a musical instrument to a computing device to allow a user to play a musical instrument in conjunction with a multimedia presentation

Disclosed is an interface device to couple a musical instrument to a computing device to allow a user to play a musical instrument in conjunction with a multimedia presentation. The interface device comprises a processor, a D/A converter, and a digital audio interface. The computing device creates a processed digital audio signal of the musical instrument based upon an original digitized audio signal of the musical instrument from the interface device. The D/A converter then converts a mixed digital signal of both the processed digital audio signal of the musical instrument and a digital audio file received from the computing device into a mixed analog audio signal. The processor controls the digital audio interface such that the mixed digital signal is transmitted through the D/A converter, and ultimately through an analog sound device, to the user.

Wireless audio and burst communication link with directly modulated THz photoconductive antenna

We demonstrate transmission of audio and burst signals through a prototype THz analog communication link employing laser-gated low-temperature-grown GaAs dipole antenna as THz emitter and receiver. The transmission distance is about 100 cm. By using a direct voltage modulation format, we successfully demodulated a burst signal with a rising time of 41 micros. The corresponding modulating bandwidth achieved was 23 kHz in this first experiment. Noise analysis reveals a 10% power fluctuation in the received signal with on-off extinction ratio of greater than 1000. The transmission of a six-channel analog and burst audio signal with least distortion is also demonstrated. We further demonstrate the fidelity of the transmission of a melody through the THz link with and without any amplification.

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A serial connection technique of speakers for multi-channel audio systems

This paper proposes a serial transmission technique of signals for multi-channel audio systems. The technique converts analog audio signals of multiple channels of an audio unit into digital signals, and transfers the digital signals to the speakers through a serial line sequentially. Each speaker is allowed to select its corresponding digital audio signals and to transform the signals into analog audio signals to make sounds for the speaker. The functions of the proposed technique are modeled in VHDL, implemented with FPGA chips, and tested for actual audio systems.

Multi-mode speaker operating from either digital or analog sources

A speaker system that detects what type of audio signal is supplied to it and automatically adjusts to reproduce sound from either an analog audio input signal or a digital audio input signal. The speaker system has one or more converters to convert each form of digital audio into analog audio, a detection means to detect whether an analog audio input signal or one of the possible digital audio signals is present and a multiplexer to switch between either the analog audio input signal or one of the outputs of the digital audio converters. Amplifier and speaker drivers are shown for completeness.

Speech band division decoder

A volume control unit 3 obtains, from a scale factor updating unit 2 , data about how much the playback scale factor of sub-bands adjacent to a sub-band to be emphasized has been reduced, and increases the analog audio signal output level of a volume control 77 to an extent corresponding to the playback scale factor necessary for restoring the playback scale factor before the reduction. As a result, the signal level of the desired sub-band (i.e., sub-band to be emphasized) is increased to obtain a sufficient sense of emphasis of the desired sub-band.

DSP architectures

As far as the future of communication is concerned, we have seen that there is great demand for audio and video data to complement text. Digital signal processing (DSP) is the science that enables traditionally analog audio and video signals to be processed digitally for transmission, storage, reproduction and manipulation. In this paper, we will explain the various DSP architectures and its silicon implementation. We will also discuss the state-of-the art and examine the issues pertaining to performance.

Method and circuit for decoding an analog audio signal using the BTSC standard

An integrated circuit for decoding an analog audio signal includes a tuner that receives the analog audio signal and provides a sound intercarrier frequency signal. A digital demodulator receives and digitizes the sound intercarrier frequency signal to provide a digitized sound intercarrier frequency signal, and digitally demodulates the digitized sound intercarrier frequency signal to provide a digitized multichannel television sound (MTS) demodulated signal. A digital broadcast television system committee (BTSC) compatible decoder receives and decodes the digitized multichannel television sound (MTS) demodulated signal, and provides a summed (L+R) audio output signal and a difference (L−R) audio output signal.

Method for processing and reproducing audio signal

A digital audio signal obtained from an analog audio signal indicating a music, pieces of audio reproduction control information respectively indicating the adjustment of a sound quality of the music are added to the digital audio signal, and the digital audio signal is recorded with the pieces of audio reproduction control information as packed data. When a user selects a piece of particular audio reproduction control information from the pieces of audio reproduction control information after the packed data is read out, levels of pieces of audio data of the digital audio signal are adjusted according to the particular audio reproduction control information, so that the user can entertain the music at a desired sound quality.

Circuit for decoding an analog audio signal

An integrated circuit for decoding an analog audio signal includes a tuner that receives the analog audio signal and provides a sound intercarrier frequency signal. A digital demodulator receives and digitizes the sound intercarrier frequency signal to provide a digitized sound intercarrier frequency signal, and digitally demodulates the digitized sound intercarrier frequency signal to provide a digitized multichannel television sound (MTS) demodulated signal. A digital broadcast television system committee (BTSC) compatible decoder receives and decodes the digitized multichannel television sound (MTS) demodulated signal, and provides a summed (L+R) audio output signal and a difference (L−R) audio output signal.

An analog record, playback, and processing system on a chip for mobile communications devices

We present a low-power chip for nonvolatile storage and conditioning of voiceband analog audio signals and nonvolatile storage for use in mobile communication devices. The system allows for direct interface to acoustic transducer elements and provides signal conditioning to gain adjust, multiples, filter, and mix two independent signals. The system can record these processed signals as analog samples in a nonvolatile flash EEPROM array for later retrieval. Together with an integrated signal path, the system can store up to 8 minutes of audio signal. Control of the system is achieved via a serial interface, which is used to configure and control the device. All necessary components of the system are provided on chip including analog processing elements, nonvolatile storage, high voltage, and reference generation.

An Intelligent Dynamic Reconstruction Filter for Audio Signal Reconstruction using Neural Networks

The goal of any digital audio system is to sample and reconstruct an analog audio signal, without noticeable changes to the original signal. Currently, two major types of reconstruction filters, brickwall and monotonic filters, are used to smooth a sampled analog audio signal during its reconstruction. Brickwall filters work best on the reconstruction of smooth signals, and the monotonic filters are best for the reconstruction of transient signals. Since audio is composed of mixed transient and smooth signals, both of these filters will introduce undesirable artifacts to the signal during its reconstruction. This paper presents a new neural-network-based dynamic reconstruction filter that can change its behavior to best match the type of signal that is being filtered.

Digital audio signal recording device for motion picture film and audio signal recording system for motion picture film employing the digital audio signal recording device

A digital audio signal recording device which, in conjunction with a film magazine loaded with a motion picture film and an analog audio recording device recording analog audio signals on the motion picture film, constitutes an audio signal recording system for the motion picture film capable of recording both analog audio signals and digital audio signals on the motion picture film delivered from the film magazine. The digital audio signal recording device transports the motion picture film, delivered from the inside of the film magazine via the film reel-out section, into the digital audio signal recording device via a first film feed-in section, and then transports the film to the analog audio signal recording device via a first feed-out section and a film lead-in section. The motion picture film introduced into the analog audio signal recording device is returned into the digital audio signal recording device via a film lead-out section and a second film feed-in section. The digital audio signals are recorded in the digital sound track area on the motion picture film returned back into the digital audio signal recording device by a digital audio signal recording section provided in the digital audio signal recording device. The motion picture film, on which the digital audio signals have now been recorded, is delivered via a second film feed-out section towards the film magazine so as to be taken up in the film magazine by a film taker-in section.

Simultaneous recording of analog signals, audio and video using a VCR

An interface to record analog signals from DC to 20 kHz on the stereo sound channels of video cassette recorders is described. This interface is an effective solution for the implementation of experiments requiring simultaneous recording of electrophysiological analog data, audio and video signals, since all these data are recorded on the same tape. The system requires the construction of a simple analog interface and minor modifications internal to the VCR, required to bypass non-linear elements present on the audio signal path. The particular solution used to adapt for Sony VCR model SLV-575UC is given in full detail. The general requirements the other VCR makes and models must fulfil to make the adaptation feasible and guidelines for the design of interfaces are discussed.

Digital audio amplifier having a high power output level and low distortion

A digital amplifier having a modulator for transforming the analog audio input signal into two complimentary trains of pulse width modulated signals for driving the power switches, which modulator utilizes a precision triangular waveform to control the transformation. The modulator employs all differential processing without saturated transistors to achieve a minimal distortion modulation system. Upon the occurrence of over-modulation, pulses from the width modulated pulse train produced by the modulator are dropped, and the resulting loss of pulses is detected and fed back to a variable gain amplifier stage of the system. The distortion detection system for detecting the missing pulses utilizes a timing network to establish the missing pulse period necessary to begin input attenuation of the audio signal by the variable gain amplifier stage. A reconstruction filter includes a low-pass section for recovering the amplified audio and a notch filter section having notches centered to eliminate the fundamental of the switching frequency. A variable inductor is employed in connection with a section of the notch filter for aiding in the proper notching of the signals to maximize suppression of the switching frequency, such inductor being coupled to the feedback network, and is adjusted by the operator while observing output distortion until the distortion attains a minimum level. The reconstruction filter also includes a tapped inductor functioning as an autotransformer which introduces offsets in the output switching waveform of the power amplifier to effectively compensate for crossover distortion due to energy recovery diode overswings and power switch voltage drops.

Computer simulation of bias recording. II. Saturation output level

For pt.I see ibid., vol.24, p.2235-46, Sept. 1988. Magnetic recording involving the storage of analog audio signals is investigated by computer simulating the write-read process. Using the magnetic and technical data of the recording system, the calculations result in data on output voltages which are in quantitative agreement with corresponding experimental data. As a concrete example for an application of the model the recording of high-frequency audio signals on cassette recorders is analyzed. The influence and the importance of various media and system data on the saturation output level at 10 kHz are determined on the basis of computer simulation. >

Speech sampling and companding device

An analog speech signal is sampled of a nominal rate of 6 kilohertz and digitized in a Mu-Law Encoder. The digital output of the Mu-Law Encoder is converted by a microprocessor performing table look-up to linearized pulse code modulation (PCM) samples nominally of eight bits per sample. Using a BSPCM (Block Scaled Pulse Code Modulation) method, in each block of nominally 246 eight-bit PCM samples (representing approximately 41 milliseconds), the maximum and minimum sample values are found and used to calculate a scale factor equal to the maximum sample value minus the minimum sample value, with the difference being then divided by a constant number nominally equaling 16. Then the BSPCM samples are generated from the PCM samples each as a corresponding one PCM sample minus the minimum PCM sample value, the difference being then divided by the scale factor. In effect, the bit rate is reduced by adjusting the step size to follow the local block dynamic range. The BSPCM samples so created are susceptible to signal processing operations like as to PCM samples. When the BSPCM encoded words plus the minimum, PCM encoded, sample plus the range increment scale factor are stored as a data block, then such data block can, at a later time, be decoded, or reconstituted, into linear PCM data. A silence interval is encoded as zero amplitude using run length coding of the number of blocks. Such digital PCM data can be converted to an analog audio signal for voice output across a telephone system.

Reverberator having tapped and recirculating delay lines

A reverberator comprises a tapped delay line (2, 3, 4, 5, 10) connected to an analog audio signal source (1) for deriving therefrom at least one pair of output signals which are respectively delayed by first and second different values with respect to the source signal, the ratio of the first to second values being an irrational number. A recirculating delay line (6, 7, 11, 12) is connected to the output of the tapped delay line having a delay element (63, 73, 113, 123) for introducing an additional delay to the output signals of the tapped delay line and a resistive recirculating path (65, 75, 115, 125) for recirculating the additionally delayed signals through the delay element. The output of the recirculating delay line is combined with the source signal to derive a reverberating audio signal.

Digital Audio Modulation in the PAL and NTSC Laservision Video Disc Coding Formats

We present an extension of the current LaserVision video disc format that includes a digital audio signal. We show the feasibility of a combined digital audio signal according to the Compact Disc Digital Audio format and the current analog audio signals in the NTSC video format, enabling the realization of a compatible system. For the PAL and SECAM video formats we show the feasibility of digital audio, but unfortunately it cannot be combined with the analog audio carriers.