An improved method for compressing a cyclic analog signal in which adjacent positive and negative half cycles are similar, such as an audio signal, and for reconstructing or synthesizing a new analog signal from the compressed signal. The original signal is compressed by removing one of the positive or the negative half cycles and, in a preferred embodiment, digitizing the remaining half cycles. The new analog signal is generated by sequentially applying each half cycle to an output and simultaneously storing such applied half cycle in a memory. After a half cycle is applied to the output and prior to applying the next half cycle in the sequence, the stored half cycle is applied to the output at a reversed polarity to simulate the portion which was originally removed.
An integrated circuit module in which an error detection circuit compares data generated internally on module with data generated externally from another substantially identical module. An error detect output is asserted upon the condition that data generated internally on module and data generated externally from module do not match. A circuit alters the internally generated data by injecting a zero bit and then a one bit data into the internally generated data to thereby generate altered data. Error anticipation control logic generates a test condition, which corresponds to the expected error condition caused by altered data, by first expecting to detect the effect of the injected zero bit and then expecting to detect the effect of the injected one bit. An error-0 comparison circuit compares the actual error detect output with expected error detect output for the zero bit. An error-1 comparison circuit compares the actual error detect output with expected error detect output for the one bit. An error output is asserted if the actual error detect output and the expected error detect output do not match in either of the two cases.
In a time division multiplexing transmission system for transmitting a plurality of interrupted signals such as picture signals and pulse modulated signals corresponding to a continuous signal such as PCM audio signals through a transmission path having a plurality of channels, each of which alternately has pause and signal periods at a predetermined time sequence, the interrupted signals are allotted to the pause period and the pulse modulated signals are alloted to the signal period. The pulse modulated signals are divided into two series of signals, each of which is divided into first signal parts having a period equal to the signal period and second signal parts having a period equal to the pause period. The first or second signal parts of both signal series are delayed in such a way that the delay means in the receiver can be simplified and the delayed first or second signal parts and the non-delayed second or first signal parts are combined in a way that these signal parts form a new interrupted signal series allotted in the signal periods. At the receiver the received signal is converted to the original pulse modulated signals so as to reproduce the original continuous signal. This system particularly suggests a novel audio signal transmission with high quality of reproduced sound in the case of intermittently transmitting a PCM audio signal together with a video signal, such as a still picture transmission system in which the audio and video signals are transmitted per unit time, i.e., audio and video frames. In addition, this system has an advantageous effect in which the construction of the receiver can be simplified.
Left right and surround components of a stereo signal are coded into a monaural and small co-channel providing volume steering for recreating a stereo effect with a substantially reduced bit rate. The signal is split into a sum and difference signal, the difference signal is randomized and the sum is added to the randomized difference to comprise the single audio channel. A functional relationship is solved for left and right volumes which is then transmitted for intervals on the co-channel. Decoding of the single transmission channel directs it to left, right, or surround channels based on decoding on the logic co-channel. The co-channel updates left and right volume levels which are interpolated through time to effect smooth volume change. Surround gain is determined from left and right channel gains to maintain unity total volume, with the sum of the squares of the three volume controls being unity.
A method and apparatus for analyzing and synthesizing speech information in which a predetermined vocabulary is spoken into a microphone, the resulting electrical signals are differentiated with respect to time, digitized, and the digitized waveform is appropriately expanded or contracted by linear interpolation so that the pitch periods of all such waveforms have a uniform number of digitizations and the amplitudes are normalized with respect to a reference signal. These "standardized" speech information digital signals are then compressed in the computer by subjectively removing and discarding redundant speech information such as redundant pitch periods, portions of pitch periods, redundant phonemes and portions of phonemes, redundant amplitude information (delta modulation) and phase informaton (Fourier transformation). The compression techniques are selectively applied to certain of the speech information signals by listening to the reproduced, compressed information. The resulting compressed digital information and associated compression instruction signals produced in the computer are thereafter injected into the digital memories of a digital speech synthesizer where they can be selectively retrieved and audibly reproduced to recreate the original vocabulary words and sentences from them.
Sound monitoring apparatus is described which provides a slowed down version of the original sound, e.g. a heart beat without changing the quality of the sound as perceived by a trained listener such as a physician. The amount by which the sound is slowed down can be varied by the user. The original sound is converted to an analog electrical signal, digitized and electronically processed in a microprocessor-based circuit such that digital data corresponding to cycles of the analog signal are stored in a random access memory. The microprocessor reads out the digital data a predetermined number of times corresponding to a number of sound cycles according to the rate set by the physician. The digital data is reconstituted into sound which is composed of replicated sets of cycles of the original sound. The repetition rate of the heart-beat in the output sound is a fraction of the repetition of the heart-beat in the original sound but the sound quality appears the same to the physician as the pitch is invariant. In a preferred embodiment stored digital data corresponding to each input half signal is duplicated and data corresponding to the duplicated positive half-cycle is inverted to give data corresponding to a negative half-cycle and the non-duplicated negative half-cycle data is inverted to give a positive half-cycle data. In another embodiment, the signal processing circuitry is incorporated into a conventional stethoscope and the circuitry is implemented using low power CMOS technology. The invention has application in other fields such as vibration analysis.
