Motorola Corporation

Radu C Frangopol

Marc Andre Boillot

Jean Khawand

Wayne W Ballantyne

Ali Behboodian

Audley F Patterson

Adarsh S Desai

A sigma delta digital-to-analog (D/A) converter system () includes a summing device () at an input of a D/A converter (), and a low frequency low amplitude wave signal () injected at an input of the summing device that remains unfiltered and is used to suppress spurious tone artifacts. The D/A converter system can further include an amplitude control and a frequency control for selectively adjusting the frequency and the amplitude of the low frequency low amplitude wave signal being injected. Note, the low frequency low amplitude repeating wave signal generator can take the form of a digital signal processor (DSP) () having the appropriate software to generate such signals.

Method and system for mitigating background noise for a Sigma-Delta digital-to-analog converter

Systems (or) and methods (or) are provided for selecting a post-compression waveform from a post-compression waveform table () and supplying it to a synthesis engine (). The post-compression waveform is based upon a set of post-compression coefficients determined by generating a frequency-domain representation of a periodic signal, the representation including at least one pre-compression frequency-domain sample (), and performing a threshold-based compression of the pre-compression frequency-domain samples. Systems and methods also include indexing and storing () post-compression coefficients in a post-compression coefficient table (), generating () a post-compression waveform based upon the set of post-compression coefficients, and placing () the post-compression waveform in the table prior to the selecting (). The system and method also include performing () a read-ahead operation on a sound file before selecting the post-compression waveform, the read-ahead operation indicating the post-compression waveform to be selected and supplied to the synthesis engine.

System and methods for memory-constrained sound synthesis using harmonic coding

A method of scaling polyphony can include identifying music data to be played (), wherein the music data indicates instruments to be used and each instrument has an assigned priority. A measure of polyphony needed to play the music data can be compared with polyphony of a sound generating device (). If the measure of polyphony exceeds the polyphony of the sound generating device, the music data can be played without using one or more instruments indicated by the music data according to the assigned priorities ().

MIDI scalable polyphony based on instrument priority and sound quality

A system, method and computer readable medium for adjusting volume levels of a Musical Instrument Digital Interface (MIDI) sound file for optimizing play on a sound device. The method on an information processing system includes calculating a first set of loudness levels for each instrument in a MIDI sound file and calculating a second set of loudness levels corresponding to an audio output range of a sound device. The method further includes generating a mapping between the first set of loudness levels and the second set of loudness levels corresponding to the audio output range of the sound device. The method further includes generating a gain term for each note in the MIDI sound file and modifying the MIDI sound file using the second set of loudness levels and the gain term for each note in the MIDI sound file.

Balancing MIDI instrument volume levels

A percussive sound contains both harmonic and non-harmonic frequency spectral content. To reproduce a particular percussive sound, such as the sound of a drum or cymbal or hand clap, for example, the harmonic and non-harmonic content are determined empirically. Also, and tendency of the harmonic content to change over time, and the temporal aspects of attack, sustain, and decay are likewise determined empirically. These attributes are represented in the invention in a percussive sound file which includes a harmonic content profile ( ), noise shape filter ( ), Doppler shift profile ( ), and a time wave shaping profile ( ). The harmonic content profile is used by an FM generator ( ) to generate a frequency modulated signal ( ). The noise shape profile is used by a noise generator ( ) to generate and shape the non-harmonic spectral content. While the sound is being generated, the Doppler shift profile is used to adjust the base frequency of the FM signal. The harmonic and non-harmonic signals are scaled ( ) and summed ( ). The summed signal is then shaped ( ) in time to substantially simulate the attack, sustain, and decay properties of the sound. The shaped, summed signal is then played by an audio circuit and converted to an acoustic signal.

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Plantation, FL, United States of America

Radu C Frangopol