Method and apparatus employing audio frequency offset extraction and floating-point conversion for digitally encoding and decoding high-fidelity audio signals

A decoder for decoding an audio signal having been represented by an initial series of pulse-code modulated (PCM) data values occurring at a first rate, said audio signal including low frequency components of high amplitude and high frequency components of relatively low amplitude, said audio signal having been encoded by:

• (a) extracting from said PCM data a series of representative values occurring at a second rate substantially lower than said first rate, half of said second rate being at an intermediate frequency in the audio spectrum, said intermediate frequency lying between the frequencies of said low frequency components and said high frequency components;
• (b) offsetting said PCM values in accordance with corresponding values in said series of representative values to obtain a series of adjusted PCM data values;
• (c) extracting a series of scale factors from said series of adjusted PCM data values, said scale factors occurring at a rate substantially less than said first rate, said series of scale factors being selected in accordance with the magnitudes of said adjusted PCM data values;
• and (d) scaling said adjusted PCM data values by corresponding ones of said scale factors, to obtain a scaled series of PCM data values, so that the combination of said series of representative values, said series of scale factors and said scaled series of PCM data values encode said audio signal at a substantially lower data rate than said initial series of PCM data values;
• said decoder comprising:
• (a) means for receiving said series of representative values, said series of scale factors, and said series of PCM data values;
• (b) means for translating the scaled PCM data values in accordance with said scale factors to obtain a series of translated PCM data values;
• and
• (h) means for combining corresponding ones of said representative values with said translated PCM data values to obtain a series of PCM data values approximating said initial data values, and thereby preventing the low frequency components of high amplitude from having a destructive effect upon the high frequency components of relatively low amplitude.

A method of decoding an audio signal which has been digitally encoded from an initial series of pulse-code modulated (PCM) data values occurring at a first rate, said audio signal including low frequency components of high amplitude and high frequency components of relatively low amplitude, said method including the steps of:

• (a) extracting from said PCM data values a series of representative values occurring at a second rate substantially lower than said first rate, half of said second rate being at an intermediate frequency in the audio spectrum,
• (b) offsetting said PCM data values in accordance with corresponding values in said series of representative values to obtain a series of adjusted PCM data values, and
• (c) converting the adjusted PCM data values to a series of floating-point data values by extracting exponents, so that the combination of said series of representative values and said series of floating-point data values encode said audio signal at a substantially lower-data rate than said initial series of PCM data values,
• said method of decoding comprising the steps of:
• (d) translating the series of floating-point data values to obtain a series of fixed-point data values, and
• (e) combining corresponding ones of said representative values with said fixed-point data values to obtain a series of PCM data values approximately said initial series, and thereby preventing the low frequency components of high amplitude from having a destructive effect upon the high frequency components of relatively low amplitude.

A method for digitally encoding an audio signal represented by an initial series of fixed-point pulse code modulate (PCM) data values occurring at a predetermined sampling rate, each value being represented by a predetermined number of bits, said audio signal including low frequency components of high amplitude and high frequency components of relatively low amplitude, said method comprising the steps of:

• dividing said series of fixed-point PCM data values into blocks, each block comprising a plurality of consecutive PCM data values, each block including a predetermined number of said consecutive PCM data values so that said blocks occur at a determined block rate being less than said sampling rate, half of said block rate being at an intermediate frequency in the audio spectrum,
• centering the fixed-point PCM data values within each block about a zero reference level by extracting a common offset value K for the data values within the block,
• selecting for each block a respective one of a plurality of predetermined scale factors, the respective scale factor for each block being selected in accordance with the centered fixed-point PCM data value of maximum magnitude in the block,
• scaling the centered fixed-point PCM data values in each block by the respective scale factor selected for the block to obtain a series of scaled PCM data values that are represented by a predetermined number of bits less than the number of bits of the PCM data values in the initial series, so that the combination of the series of floating-point PCM data values, the common offsets K for the blocks, and the common scale factors for the blocks encode said audio signal at a substantially lower bit rate than said initial series of fixed-point PCM data values, and thereby preventing the low frequency components of high amplitude from having a destructive effect upon the high frequency components of relatively low amplitude.

A method for digitally encoding an audio signal represented by an initial series of fixed-point pulse code modulated (PCM) data values occurring at a predetermined sampling rate, each value being represented by a predetermined number of bits, said audio signal including low frequency components of high amplitude and high frequency components of relatively low amplitude, said method comprising the steps of:

• dividing said series of fixed-point PCM data values into blocks, each block comprising a plurality of consecutive PCM data values, each block including a predetermined number of said consecutive PCM data values so that said blocks occur at a predetermined block rate being less than said sampling rate, half of said block rate being at an intermediate frequency in the audio spectrum,
• centering the fixed-point PCM data values within each block about a zero reference level by extracting a common offset value K for the data values within the block,
• transforming the centered fixed-point PCM data values within each block to a floating-point format in which the centered PCM data values are represented by a predetermined number of bits less than the number of bits of the PCM data values in said initial series, and in which a common exponent is determined for the block corresponding to a common scale factor for the floating-point conversion, so that the combination of the series of floating-point PCM data values, the common offsets K for the blocks, and the common exponents for the blocks encode said audio signal at a substantially lower bit rate than said initial series of fixed-point PCM data values, and thereby preventing the low frequency components of high amplitude from having a destructive effect upon the high frequency components of relatively low amplitude.

A method for decoding an audio signal which has been encoded from an initial series of fixed-point pulse code modulated (PCM) data values occurring at a predetermined sampling rate and representing samples of said audio signal, each value being represented by predetermined number of bits, said audio signal including low frequency components of high amplitude and high frequency components of relatively low amplitude, said audio signal having been encoded by:

• dividing said series of fixed-point PCM data values into blocks, each block comprising a plurality of consecutive PCM data values, each block including a predetermined number of said consecutive PCM data values so that said blocks occur at a predetermined block rate being less than said sampling rate, half of said block rate being at an intermediate frequency in the audio spectrum,
• centering the fixed-point PCM data values within each block about a zero reference level by extracting a common offset value K for the data values within the block, and
• transforming the centered fixed-point PCM data values within each block to a floating-point format in which the centered PCM data values are represented by a predetermined number of bits less than the number of bits of the PCM data values in said initial series, and in which a common exponent is determined for the block corresponding to a common scale factor for the floating-point conversion, so that the combination of the series of floating-point PCM data values, the common offsets K for the blocks, and the common exponents for the blocks encode said audio signal at a substantially lower bit rate than said initial series of fixed-point PCM data values,
• said method of decoding comprising the steps of:
• receiving said common offset values K, said common exponent values, and said floating-point PCM data values,
• transforming the received floating-point PCM data values in accordance with their respective received exponent values to recover fixed-point PCM data values,
• de-centering the fixed-point PCM data values by adding to them their respective common offset values, and
• converting the de-centered fixed-point PCM data values to analog form, thereby preventing the low frequency components of high amplitude from having a destructive effect upon the high frequency components of relatively low amplitude.

A decoder for decoding an audio signal which has been encoded from an initial series of fixed-point pulse code modulated (PCM) data values occurring at a predetermined sampling rate and representing samples of said audio signal, each value being represented by a predetermined number of bits, said audio signal including low frequency components of high amplitude and high frequency components of relatively low amplitude, said audio signal having been encoded by:

• dividing said series of fixed-point PCM data values into blocks, each block comprising a plurality of consecutive PCM data values, each block including a predetermined number of said consecutive PCM data values so that said blocks occur at a predetermined block rate being less than said sampling rate, half of said block rate being at an intermediate frequency in the audio spectrum,
• centering the fixed-point PCM data values within each block about a zero reference level by extracting a common offset value K for the data values within the block, and
• transforming the centered fixed-point PCM data values within each block to a floating-point format in which the centered PCM data values are represented by a predetermined number of bits less than the number of bits of the centered PCM data values, and in which a common exponent is determined for the block corresponding to a common scale factor for the floating-point conversion, so that the combination of the series of floating-point PCM data values, the common offsets K for the blocks, and the common exponents for the blocks encode said analog audio signal at a substantially lower bit rate than said initial series of fixed-point PCM data values, and thereby limiting the normally destructive effect that the low frequency components of high amplitude have upon the high frequency components of low amplitude,
• said decoder comprising, in combination,
• means for receiving said common offset values K, said common exponent values, and the floating-point PCM data values,
• means for transforming the received floating-point PCM data values in accordance with their respective received exponent values to recover fixed-point PCM data values,
• means for de-centering the fixed-point PCM data values by adding their respective common offset values, and
• means for converting the de-centered fixed-point PCM data values to analog form.

A method of digitally encoding, transmitting, and decoding an audio signal represented by an initial series of pulse-code modulated (PCM) data values occurring at a first rate, said audio signal including low frequency components of high amplitude and high frequency components of relatively low amplitude, said method comprising the steps of:

• (a) extracting from said PCM data a series of representative values occurring at a second rate substantially lower than said first rate, half of said second rate being at an intermediate frequency in the audio spectrum, said intermediate frequency lying between the frequencies of said low frequency components and said high frequency components,
• (b) offsetting said PCM values in accordance with corresponding values in said series of representative values to obtain a series of adjusted PCM data values,
• (c) extracting a series of scale factors from said series of adjusted PCM data values, said scale factors occurring at a rate substantially less than said first rate, said series of scale factors being selected in accordance with the magnitudes of said adjusted PCM data values,
• (d) scaling said adjusted PCM data values by corresponding ones of said scale factors, to obtain a scaled series of PCM data values, so that the combination of said series of representative values, said series of scale factors and said scaled series of PCM data values encode said audio signal at a substantially lower data rate than said initial series of PCM data values,
• (e) transmitting said series of scaled PCM data values, said series of representative values and said series of scale factors over a band-limited channel,
• (f) receiving said series of scaled PCM values, said series of representative values and said series of scale factors from said band-limited channel,
• (g) translating the scaled PCM data values in accordance with said scale factors to obtain a series of translated PCM data values, and
• (h) combining corresponding ones of said representative values with said translated PCM data values to obtain a series of PCM data values approximately said initial series, and thereby preventing the low frequency components of high amplitude from having a destructive effect upon the high frequency components of relatively low Amplitude.