Analog Audio Research Articles

Audio broadcast receiving apparatus and method

An audio broadcast receiving apparatus and an audio broadcast receiving method implemented by the audio broadcast receiving apparatus are disclosed. A tuner simultaneously receives a radio wave in a specific frequency band of the analog audio broadcast and a radio wave of the digital audio broadcast transmitted in parallel with the analog audio broadcast in the specific frequency band. A demodulator retrieves the intended audio information by demodulating the signal received by the tuner. An analog receiving signal bandpass filter selectively passes the receiving signal corresponding to the analog audio broadcast radio wave. An analog receiving signal amplifier outputs an output signal representing an amplified receiving signal that has passed the analog receiving signal bandpass filter to the demodulator, and the gain of the amplifier is controlled in accordance with the level of the same output signal. A digital receiving signal bandpass filter selectively passes the receiving signal corresponding to the digital audio broadcast radio wave. A digital receiving signal amplifier amplifies the receiving signal that has passed through the digital receiving signal bandpass filter with a gain controlled in accordance with the level of the same output signal and outputs the amplified signal to the demodulator.

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.

Transitioning from analog to digital audio recording in childhood speech sound disorders

Few empirical findings or technical guidelines are available on the current transition from analog to digital audio recording in childhood speech sound disorders. Of particular concern in the present context was whether a transition from analog‐ to digital‐based transcription and coding of prosody and voice features might require re‐standardizing a reference database for research in childhood speech sound disorders. Two research transcribers with different levels of experience glossed, transcribed, and prosody‐voice coded conversational speech samples from eight children with mild to severe speech disorders of unknown origin. The samples were recorded, stored, and played back using representative analog and digital audio systems. Effect sizes calculated for an array of analog versus digital comparisons ranged from negligible to medium, with a trend for participants' speech competency scores to be slightly lower for samples obtained and transcribed using the digital system. We discuss the implications of these and other findings for research and clinical practise.

What exactly is VoIP?

Voice over Internet Protocol (or VoIP) is on the verge of revolutionising the telecommunications industry. It is a totally new way of carrying voice traffic, and one that telecommunications companies will have no choice but to fully embrace if they are to remain competitive. Until recently, all voice traffic has been carried using what is called circuit switched technology in which a physical circuit is switched to provide a call for a user. The radically new approach owes much to the overwhelming success of the Internet, which paved the way for the adoption of packet data communications based on Internet Protocol (IP). The concept of voice over IP is quite straightforward. An endpoint - which may be a VoIP phone or a computer - consists of a few blocks. A vocoder converts the analogue audio to and from a digital format. The data generated is split into packets and sent, with the relevant protocol, into the outside world via a network interface card. Signalling and call control are also applied through this card. The article outlines the the basic operation of VoIP.

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.

Second-harmonic magnetoresistive imaging to authenticate and recover data from magnetic storage media

A scanning magnetoresistive (MR) microscope is developed for high-resolution imaging of magnetic tapes and digital media. Second-harmonic detection is used to remove thermal anomalies. We are able to image sufficient lengths of tape for authentication purposes and for data recovery from damaged samples. The second-harmonic technique provides high-contrast magnetic images of the magnetic storage media using high-resolution MR sensors from commercial hard disk drives. These images can be directly converted into the originally recorded analog audio waveforms or digital data.

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.

A single-chip text-to-speech synthesis device utilizing analog nonvolatile multilevel flash storage

A single-chip solution for text-to-speech synthesis is presented. The chip utilizes analog nonvolatile multilevel storage (MLS) to store a corpus of natural speech elements. Synthesis is achieved by concatenating speech elements together at the word, subword, syllable, or phoneme level. An embedded micro-controller with custom peripherals performs the algorithmic tasks of the text-to-speech synthesis, namely text normalization, phonetic translation, and corpus phoneme selection. The chip is controlled via an SPI serial bus and synthesized speech is available as both analog and digital PCM audio. The chip has been fabricated in 0.5-μm flash CMOS.

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.

Antique Audio-Video in a Digital Age

A FRIEND recently asked, can I get music from my old big band LPs onto audio CDs? The question could just as well have been, can I transfer all my old children's records to CDs? This and similar queries about antique audio and video formats prompted me to begin thinking about how I would do this on my new Mac G4. And, since PC users might also need answers to such questions, I began to think about them, too. Although it's impossible to cover such a diverse topic fully in a single column, I think I can offer enough guidance to get you started. New Macs and PCs - as they are usually configured - have one very significant difference. Most new Macs have two FireWire ports (also called IEEE 1384 or I-link) and don't usually come with ports. Why not is anybody's guess. You can easily connect a newer-model video camcorder with FireWire to a Mac, and that's nice. But connecting analog sources, such as an old cassette player, requires an add-on device. Most new PCs come with an audio-in and mic-in port, but they usually lack FireWire. Neither Macs nor PCs typically come with analog video-in ports nowadays. The easy solution to a Mac's lack of an audio-in or a mic-in port is to order a device like an iMic audio adapter for $35 (plus $10 shipping) from Griffin Technology (www.griffintechnology.com). This handy little device plugs into a USB port on the Mac and has a 18-inch stereo input for audio input from such devices as microphones or audiotape players. Even if you don't plan to do the things outlined in this column, the iMic is still a great device to have. It also gives you a spare sound- port. The easy solution to the missing FireWire port on a PC is to simply order your PC with a FireWire card or add such a card at a later time. (It's less than $100.) Since PCs typically have an audio-in or mic jack, no fix is needed. How you actually connect an old analog audio device to your computer will vary, but you typically need a Y cable or adapter with male RCA plugs on the two ends of the Y and a single stereo male plug on the other end. You might also need a pair of RCA extension cables, depending on how closely you can physically place your computer and tape player. Then you plug the two RCA jacks into the line-out or play jacks on the back of your old tape deck and plug the stereo mini-pin end into the appropriate jack on your PC or into your newly purchased iMic box on your Mac. These adapters are readily available from any electronics store, such as Radio Shack, and they aren't expensive. Many people probably already have these adapters since they are the same kind of connectors used to plug portable CD players into home stereo systems. If you want to use your old turntable, things get more complicated. The audio output of a turntable needs some amplification before you can connect it to a computer. Typically this is done by connecting the turntable to a stereo system and then coming from the rec-out or line- out ports to the computer. If you want a smaller installation or you don't want to tie up a stereo, you can purchase a small phono pre-amp for about $50. Don't despair. This is a lot harder to write about than it is to wire! Unless you are extremely fastidious, you'll probably need to dust off the old turntable, and you might even need to oil it and install a new needle. For an old tape deck, you'll probably want to clean the tape heads. For old records, I recommend cleaning them with a record- cleaning cloth or with a kit such as DiskWasher. Some people I know actually wash old records in the kitchen sink using lukewarm water, gentle detergent, and a soft cloth. Dry the records using a circular motion. A few real zealots then spray the surface of the old record with a lightweight oil. I have heard that they use WD-40, but I cannot say for sure. The idea is that the oil lubricates the surface and fills in minor scratches. …

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.

Response speed and optical investigation of inGaN/gaN multiple quantum well light‐emitting diodes (LED)

AbstractTransmission experiments for a local area network (LAN) using blue‐light‐emitting diodes (LEDs) with InGaN/GaN multiple quantum well (MQW) and plastic optical fibers (POFs) have been conducted. Audio analog signals have been transmitted up to 1800 kHz, which corresponds to an optical 1.5 dB bandwidth of LED response. The response speed of these diodes has been investigated by varying the operating conditions: dc bias, pulse amplitude, and pulse shape. The LED rise time is reduced from 0.26 μs to 0.19 μs by changing the electrical pulse amplitude from 0.5 V to 1.2 V, whereas the fall time (0.20 μs) does not change. On the other hand, the fall time is reduced from 0.20 μs to 0.14 μs by adding negative pulses after the applied positive pulses. It is shown that the direct‐modulation speed of these diodes is limited by the time constant associated with device capacitance. Electroluminescence and photocurrent spectra of the LEDs were also investigated; blueshifts were observed for both peaks in the spectra. © 2001 Scripta Technica, Electr Eng Jpn, 137(3): 47–51, 2001

Multicenter audiometric results with the vibrant soundbridge, a semi-implantable hearing device for sensorineural hearing impairment

The Vibrant Soundbridge, a semi-implantable hearing device for subjects with moderate to severe sensorineural hearing impairment was introduced commercially. First audiologic results are presented on 63 patients from 10 European implant centers. Hearing loss was at 0.5, 1, 2, and 4 kHz varying between 43 and 81 dB HL. The patients used the analogue audio processor, type 302. Measured sound-field gain was compared with NAL-R target values. For most patients an acceptable agreement was found. There was a subgroup of patients, however, with relatively low gain. The results suggest that this was related to the suboptimal positioning and fixation of the transducer to the incus.

An FM subcarrier data broadcast on bus transportation status indication system

FM Subsidiary Communication Authorization (SCA) has been used to broadcast background music for a long time. Analog audio programs or digital data transmission services, utilizing SCAs, such as paging, stock market and headline news updates are common in people's daily life. Some demonstrations have been offered in the past to broadcast fax messages and some studies dealing with broadcasting video programs over the FM SCA channels. This paper focuses on using the FM SCA subchannel to transmit bus service status. This application provides the transmission of accurate status of public bus transportation services in order to offer better services. The status can be displayed on bus stations display boards or on personal computer receivers.

ASTRA Digital Radio: high-quality digital audio transmissions via satellite

ASTRA Digital Radio (ADR) is a satellite audio transmission technology which has been designed to be compatible with analogue radio and television transmissions via satellite. It uses QPSK modulated subcarriers above the video spectrum. For source coding, the internationally standardized MPEG 1 Layer II digital audio compression has been selected. Owing to its backward compatibility with the existing analogue audio transmission technology, ADR provides a maximum operational flexibility for both the broadcasters and the satellite operator. For the consumer this technology allows for a very high number of near CD quality radio programmes which may be received with a low-cost receiver connected to a standard 60 cm satellite reception antenna. ADR was officially launched in August 1995 during the Internationale Funkausstellung in Berlin and currently allows for the reception of over 85 radio programmes in digital quality. Copyright © 2000 John Wiley & Sons, Ltd.

An Intelligent way of Bit-rate Reduction in Real Time Audio Systems by the use of Psychoacoustic Masking

Today analog audio is a thing of the past. With the advent of digital audio, the rush for storage space and bandwidth have been tremendous. Digital audio compression enables more efficient storage and transmission of audio data. Many forms of audio compression techniques exists offering a range of encoder and decoder complexity, myriad audio quality, and differing amounts of data compression. They range from μ-law transformation to MUSICAM algorithm. The MPEG audio standards provide a wide range of compression, audio quality and encoder/decoder complexity. This document suggests a way of achieving real time audio compression with emphasis on MPEG standards which is brilliant combination of the existing compression techniques.

Perceptual coding of digital audio

During the last decade, CD-quality digital audio has essentially replaced analog audio. Emerging digital audio applications for network, wireless, and multimedia computing systems face a series of constraints such as reduced channel bandwidth, limited storage capacity, and low cost. These new applications have created a demand for high-quality digital audio delivery at low bit rates. In response to this need, considerable research has been devoted to the development of algorithms for perceptually transparent coding of high-fidelity (CD-quality) digital audio. As a result, many algorithms have been proposed, and several have now become international and/or commercial product standards. This paper reviews algorithms for perceptually transparent coding of CD-quality digital audio, including both research and standardization activities. This paper is organized as follows. First, psychoacoustic principles are described, with the MPEG psychoacoustic signal analysis model 1 discussed in some detail. Next, filter bank design issues and algorithms are addressed, with a particular emphasis placed on the modified discrete cosine transform, a perfect reconstruction cosine-modulated filter bank that has become of central importance in perceptual audio coding. Then, we review methodologies that achieve perceptually transparent coding of FM- and CD-quality audio signals, including algorithms that manipulate transform components, subband signal decompositions, sinusoidal signal components, and linear prediction parameters, as well as hybrid algorithms that make use of more than one signal model. These discussions concentrate on architectures and applications of those techniques that utilize psychoacoustic models to exploit efficiently masking characteristics of the human receiver. Several algorithms that have become international and/or commercial standards receive in-depth treatment, including the ISO/IEC MPEG family (-1, -2, -4), the Lucent Technologies PAC/EPAC/MPAC, the Dolby AC-2/AC-3, and the Sony ATRAC/SDDS algorithms. Then, we describe subjective evaluation methodologies in some detail, including the ITU-R BS.1116 recommendation on subjective measurements of small impairments. This paper concludes with a discussion of future research directions.