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

First Page

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.

In order to provide the AES/EBU digital audio interface for the input and output interface sub-boards of the digital audio matrix equipment and improve the system integration and flexibility, a method of using FPGA to realize the multi-channel AES/EBU digital audio transmission and reception interface is designed. The sending unit receives the audio data signal in TDM format, and then converts it to AES/EBU audio format for output after demultiplexing; the receiving unit receives the AES/EBU digital audio signal, decodes it, and converts it into TDM format signal for output. The system design has passed logic function simulation and hardware testing, and the output signals of the sender and receiver are in compliance with the standard.

Upgrading Sparse NMF algorithm for blind source separation through Adaptive Parameterized Hybrid Kernel based approach

Describes a system for transmitting data packets multiplexed with digital audio and NTSC TV signals using optical fibers. Up to 18 channels are allocated for data packets transmitted at a maximum speed of 1.764 Mbps. Digital audio signals are inserted into the horizontal sync period of the NTSC TV signal. In addition to data packet transmission, the digital NTSC TV signal transmission system is used to easily distribute many types of information classified into categories. The proposed system is characterized by: firstly, this system can give a satisfactory picture quality enough to operate it together with video equipment even though the quality of the reproduced pictures is lower than that of normal digital NTSC TV picture codes obtained at a sampling frequency of 4.00/spl times/fsc with a quantizing level of 8 bits. Secondly, NTSC TV picture codes, audio codes, and data packets can satisfactorily be transmitted using inexpensive devices at a transmission rate of as low as 50 Mbps. Thirdly, this system is used for transmitting data packets and audio signal codes which are inserted in the horizontal sync period of the digital NTSC TV signal. Both data packets and audio signal codes can be distributed to the specific receivers specified by device address. A combination of edge-emitting LEDs (ELEDs) and optical fibers enable the transmission system to easily transmit a digital NTSC TV picture signal together with data packets and digital audio codes. A new approach of combining ELEDs and optical fibers to distribute CATV pictures and electronic information via the ISDN is also proposed.

Even in a conference audio system having an automatic mute release device, it is possible to reduce a delay time from an utterance to output from a speaker. The system includes: an A/D converter (33) for converting an audio signal from a plurality of microphones into a digital signal; audio level detection means to detect whether the converted digital signal level indicates an utterance or no sound; audio data storage means (32) for temporarily storing the digital signal in which the audio level detection means has detected an utterance; control means (31) for controlling storage of the audio data into the audio data storage means (32) and read out of the audio data; and a D/A converter (34) for converting the read-out audio data into an analog audio signal. When the audio level detection means has detected no sound in the series of audio data, the control means (31) hastens the audio data read-out timing in correspondence to the time of the no sound portion.

A speaker system includes a controller for driving a loudspeaker. The controller includes a sensor for detecting the present physical position of the speaker. The controller receives this position data as input, along with an audio signal to be reproduced. Using the audio signal as position data, the controller compares it with the actual sensed position data and generates an error signal. An error amplifier uses this error signal to drive the speaker, so that the speaker cone position matches the audio position defined in the audio data. In this manner, the speaker is driven by the error signal rather than the audio signal. In a preferred embodiment, the audio signal and position data are provided as digital signals, and the controller calculates the error signal in a digital signal processor.

In this paper, we propose a LED communication based multi-hop audio data transmission network system. The main contribution and features of the proposed system are as follows. First, the contribution of this research is to develope the LED communication based multi-hop transmission network system which can transmit audio data signal with long distance via multi-hops. Second, the developed system has the following features: In transmitter, audio data is transmitted after encoding with S/PDIF format via a general LED. The relay receives digital audio signal by using photo diode and then transmits the signal to receiver after error checking and amplifying. The receiver receives the encoded audio data via photo diode and then converts to analog audio signal by using decoding and amplifying. The performance evaluation of the proposed system is conducted in the laboratory with fluorescent light source. The results of the performance evaluation confirm that the system can provide high quality audio transmission from transmiter to receiver via multi-hop relays in a long distance while we can see there are differences in the transmitted audio quality according to the used LED colors.

In this paper, we propose a multi-hop transmission system using visible light communication to transmit audio data. In our proposed transmission system, at the transmitter we encode audio data based on S/PDIF standard – a popular standard for digital audio signal, and transmit the encoded audio signal via general LED. At each relay, digital audio signal is improved and amplified before sending. At the receiver, encoded audio signal from photodiode (PD) is decoded, amplified and coverted to analog audio signal. We evaluate our proposed transmission system in a room with flourescent light source. The audio signals obversed at the receiver show that with the support of relays, our proposed transmission system can provide high quality audio transmission from transmiter to receiver via multi-hop relays at a long distance.

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.

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 reconstruction of smooth signals and the monotonic filters are best for 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. The 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.

In the age of digital world, the issue of multimedia data security (such as digital audio signals, images, and videos) is a major concern due to the rapid development of digital communications and networking technologies. The available methods and algorithms to assure data protection are sincerely based on cryptography. To illustrate, the data protection methodology is inherently involved with the cryptographic primitives for secure data transmission and reception by assuming that both sides must trust on each other. Now, in the high time of modern communication technology, the range of cryptography has been drastically expanded after the development of communication means. Cryptography is essentially required to ensure data protection against penetrations and to prevent espionage. In this proposal, an illustrative conception on pulse code modulation (PCM) based audio signal encryption, undoubtedly a new technique of encryption, is given to provide a wider range of security. This very effective technique of encryption is applied in the original domain of the signal; so both encryption & decryption become much easily executable. In a brief, in this method, the audio signal samples will be encrypted using very common arithmetic algorithms & the total no of bits will be kept unaffected. So, the bandwidth required to transmit the encrypted signal will be as same as the original signal. In receiver section, the encrypted signal will be decrypted using exactly the reverse algorithm used earlier for encryption. The performance of this encryption scheme is very similar to the binary PCM system except that the audio samples are encrypted after sampling. The PCM based audio signal security system, thus, will ensure both the secure and better communication.

It is well known that, signal separation algorithms based on second order statistics alone, are not always sufficient of achieving signal separation of mixed digital modulation signals. Methods based on higher order eigenvalue or more efficiently singular value decompositions of the received sensor's covariance matrix, only work satisfactorily when the received mixed signals are linear combination of each others. In this paper, it has been shown that it is possible to retain the simplicity of the 2/sup nd/ order-based methods and still be able to successfully separate mixed digital signals, by designing the decoupling system to maximize the output normalized kurtosis functions. Next, it is proposed that in order to reduce computations and speeded up signal separation, we should perform wavelet packet decomposition of the outputs of the de-coupling network. The parameters of this network are optimally chosen through minimizing the cross correlations between wavelet packet coefficients of its outputs, at various nodes and a specific depth. Illustrative examples are given to verify these results, and demonstrate that the proposed approaches work when other methods fail.

Piano spectrum recognition system (PSRS) for the smart piano guiding framework is designed and implemented in this paper. Here, the Fourier transform method is used to analyze the difference between the frequency spectrum of the piano and other musical instruments. After analysis, it can be known that each musical instrument has its corresponding frequency multiplier amplitude, which can be recognized by extracting its timbre characteristic matrix and distinguish the timbre of different musical instruments. Also, the spectrum sensing algorithm is developed to establish a digital piano audio signal multi-source information acquisition model, and the system is implemented with the integration of efficiency estimation.

Predicting outdoor ultrafine particle number concentrations, particle size, and noise using street-level images and audio data