摘要:
Systems and methods for resizing a signal for use with a fixed-point DSP are provided. More specifically, a process called companding is used in conjunction with fixed-point devices to resize a signal to make use of the available range of these devices. In some embodiments, companding is used to improve the signal-to-noise and distortion ratio. Also, information loss associated with quantization and rounding errors can be reduced in some embodiments.
摘要:
The data acquisition portion of a data signal processing system has a nonnear data compression and conversion arrangement and a data decompression and scaling arrangement. The non-linear data compression and conversion arrangement employs graduated reference voltage levels provided by resistors having graduated, unequal values, such as values being related as a geometric progression. The graduated form of the reference voltage levels provide a form of data compression wherein large value and small value measurements will have essentially the same fractional resolution. The data decompression and scaling arrangement can decompress previously compressed digital signals and limit the digital data width of such signals in systems which have input dynamic range requirements greater than their output resolution requirements.
摘要:
A data companding method instantaneously compands the 14-bit digital audio signals to be outputted from a satellite broadcast receiver and so on is realized with the given conversion rule, which has a superior companding characteristics across the whole input range of the 14-bit data with SN being not deteriorated as compared with the 14-10 near-instantaneous companding law at the satellite broadcast transmission. A data compressor and a data expander in accordance with the present invention are realized by the construction of the combination of the coefficient table, a data shifter composed of a pluarlity of 4 to 1 data selectors, and an adder.
摘要:
In a digital text-to-speech conversion system of the type usually contained in all-software form on a floppy disk, memory requirements for the storage of digitized waveform samples are reduced while speech quality is improved, by providing compression techniques and anti-distortion techniques which interact to provide clear speech at widely varying speeds with a minimum of memory. These techniques include using Huffman coding of first- or second-order differences, encoding only differences between successive waveforms where feasible, using a demi-diphone organization of the speech to allow use of the same instruction lists for several sounds, selectively deleting or repeating waveforms in the concatenation to vary speed without affecting pitch, and encoding waveforms linearly or anti-logarithmically for storage while converting the stored linear or anti-logarithmic codes to logarithmic codes such as .mu.-law codes upon retrieval.
摘要:
A circuit for companding digital signals, incorporating a bidirectional barrel shifter in the form of an array of transmission gate cells for shifting step bits of a compressed PCM signal a predetermined number of locations to the left in response to the magnitude of chord bits thereof. The shifted bits are applied to a linear digital signal bus in the form of an expanded linear representation of the PCM signal. Linear digital signals appearing on the linear bus are compressed via shifting to the right through the barrel shifter and are applied therefrom to a PCM signal bus. The companding circuit is inexpensive, can accommodate both A-law and .mu.-law PCM protocols, is fully static and operates at high speed.
摘要:
The circuit comprises two registers (SR12, SR8) which temporarily memorize uncompressed and compressed PCM words, respectively. The uncompressed word is sequentially read and a counter (CNT) counts the number of zeroes present in the most significant positions of the absolute value of the word. The compressed word is composed, in the corresponding register, of the sign bit of the uncompressed word, followed by the counting bits of the counter and by the bits of the positions of the uncompressed word following those of the counted zeroes.
摘要:
The most significant bits of each input word of a linear PCM signal address a read-only memory (ROM). Stored at each ROM address is a table of instruction words, one instruction word for each bit in the output word. The instruction words direct a switch to select each bit for the output word from a choice of 1, 0, or an input bit PCM compression according to a predetermined characteristic is implemented by instruction words which direct selection of the proper chord bits from 1 or 0 and the proper step bits from the input bits. A different compression characteristic may be implemented by a different set of stored instruction words. The instruction words also command selective bit inversion and selective inclusion of the bit in a parity check. Implementation of the translator with commercially available hardware is described.
摘要:
To test the performance of a PCM terminal operating in the TDM mode, a simulated message or noise signal is generated by extracting a recurrent sequence of code words from a read-only memory, the stored code words representing a set of values logarithmically related to sucdessive amplitudes of a sine wave or other periodic oscillation to be reproduced. These words are read out at a fixed scanning frequency f.sub.o but with skipping of (p-l) memory stages if a signal frequency pf.sub.o is to be simulated. A volume selector generates supplemental code words which are additively or subtractively combined with the extracted code words to simulate a desired voltage level. The resulting code words are compressed in a code converter in which the quantum steps of the segments of a compander characteristic, conforming to the chosen logarithmic function, are stored in another read-only memory.
摘要:
In a PCM system, character compression using nearly instantaneous companding (NIC) is known to obtain a reduction in the number of bits transmitted. Unfortunately, in tandem digital PCM-to-NIC-to-PCM conversions, a signal delay results because a maximum segment value is usually unknown until a block of PCM characters, which are to be converted to make up a block of NIC characters, has been received. Further, if the PCM block is received out-of-phase, a signal degradation may result because the incorrect maximum sgement value may be used during a subsequent conversion. To mitigate delay, the disclosed processor discards certain least significant bits of a PCM character and substitutes therefor a predetermined bit pattern corresponding to the difference between the maximum segment value and the segment value of the first character of the block. To mitigate degradation, the processor adjusts the block phase upon detection of the absence of a predetermined difference between the maximum segment value and the segment value of another character in the block.
摘要:
A pulse code modulation compressor converts linear pulse code modulation words into companded pulse code modulation words. A read only memory is loaded with a conversion table and used to perform successive approximations for the conversion. The converted output is then passed through a double buffer to allow independent operation of the conversion clock and output clock.