System and method for generating and attenuating digital tones

FIELD OF THE INVENTION

This invention relates to the generation and attenuation of digital signals for input to a digital to analog converter to produce an audible tone. More specifically, it relates to use of a digital signal processor (DSP) to generate pulse coded modulation (PCM) values representing a set of predefined tones in a memory space and processing cycles efficient manner.

BACKGROUND OF THE INVENTION

A tone is a pure sine wave. Pulse coded modulation (PCM) data is a digital representation of an analog signal, such as a sine wave, at fixed time intervals.

Digital signal processors (DSPs) may be used to generate tones. This they do by generating electrical signals which are input to a digital to analog converter (DAC) to produce an analog electrical signal that will cause one of a set of tones to be produced with an appropriate audio amplifier and speaker. These processors usually have limited function, providing only fixed point operations and multiply, but not divide.

If a tone stops at a non-zero value, or the tone goes to zero at a high rate, or the sine is distorted by being attenuated at an increasing rate, the resulting sound will contain “clicks”, “pops”, or “thuds”. Since tone duration may be short (say, 0.1 seconds) and there may only be between 16,000 and 48,000 samples per second, the whole tone may contain only 1600 samples. Attenuation should be complete in about ten percent of these samples, and the solution should use little code and little memory.

For short tones the attenuation duration must also be short. As the duration of a sine or of a few sine oscillations approach the period of the attenuation duration, noiseless attenuation becomes difficult. Some distortion must be expected. For instance, a sine wave cannot be changed during a half wave and still be a pure sine wave.

Synchronization of digital video and digital audio data streams is a requirement of the art. Because digital video data is typically compressed on picture frames, and audio is typically compressed on frames of a fixed number of samples, synchronization following a discontinuity in the audio program has heretofore required that a certain frame boundary be identified as a sync point. There is, therefore, a need in the art for an improved method which avoids the need to re-synchronize video and audio data by allowing decode of the audio program to continue. In accordance with the present invention, this is accomplished by substituting a digital tone value for the audio program output value. This digital tone generation is an additional processing load on the DSP and it is desirable to minimize this load.

It is, therefore, an object of the invention to generate short tones with rapid attenuation while avoiding objectionable noise.

It is a further object of the invention to operate a digital signal processor in a memory space and processing cycles efficient manner to generate and attenuate tones.

It is a further object of the invention to attenuate a tone without creating, or at least minimizing, additional sounds or artifacts at the end of the tone, such as “clicks”, “pops”, or “thuds”.

It is a further object of the invention to produce a large number of tones and tone durations across and beyond the entire audio range.

It is a further object of the invention to produce a sine wave of highly accurate frequency.

It is a further objective of the invention to replace a segment of a playing audio stream with a tone of the same sampling frequency as the audio stream in order to maintain synchronization between audio and video data.

SUMMARY OF THE INVENTION

In accordance with the method of the invention, an audible tone is generated and attenuated over a wide frequency range, such as throughout and beyond the human audible range, the tone selectively being of short duration, including the steps of generating during a tone period a digital representation of the sine of a requested tone frequency and amplitude; generating during an attenuation period a digital representation of a moderately disturbed but continuous sine of decreasing amplitude; and generating during a decay period a digital representation of a continuous function which decays to zero from the zero approach point of the sine half wave.

In accordance with a further aspect of the method of the invention, the method includes during the attenuation period the steps of multiplying the amplitude value by a fractional constant at zero crossings; incrementing within zero passing zones the amplitude between subsequent samples by reduced values to further attenuate the tone and accumulate a “bank” of accumulated reductions in increments; and while approaching zero crossings the steps of generating a pure sine wave of maximum amplitude equal to the amplitude at the end of the prior quadrant; and during a decay period, the step of generating a digital representation, of a continuous function which decays exponentially to zero amplitude.

In accordance with the system of the invention, a digital signal processor is provided for generating and attenuating an audible tone over a wide frequency range, such as throughout and beyond the human audible range, the tone selectively being of short duration. Responsive to a request to generate a tone of a specified tone and sampling index, tone request logic determines an increment angle. Responsive to said increment angle and a periodic sampling interrupt, sample generation logic generates during a tone period a digital representation of the sine of a requested tone frequency and amplitude; generates during an attenuation period a digital representation of a moderately disturbed but continuous sine of decreasing amplitude; and generates during a decay period a digital representation of a continuous function which decays to zero from the zero approach point of the sine half wave.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1

is a high level system diagram of tone generation and attenuation system in accordance with the invention in an representative system environment.

FIG. 2

is a diagram illustrating a tone period, including pure tone period, attenuation period, decay period and stop period as a function of time.

FIG. 3

is a representation of an analog sine wave output from the DAC, generated from digital inputs from DSP, of FIG.

1

.

FIG. 4

illustrates a table of tone delta T values for each of plurality of sampling frequencies.

FIG. 5

is a diagrammatic representation of a constant angular increment ΔT used in generating periodic digital sine values. ΔT (radians) is a component of angular velocity ΔT/Δt (radians/second), where Δt is the time increment between samples.

FIG. 6

is a diagrammatic representation of the use of the bits of a digital representation of an angle to determine which quadratic (that is, select the coefficients to use) and the value of the independent variable for evaluating the quadratic to estimate the sine.

FIG. 7

illustrates an enumeration of possible computed values of tone indexes to octave and note.

FIG. 8

is a diagrammatic representation of tone attenuation in a sine half wave including a zero passing zone.

FIG. 9

is a diagrammatic representation of exponential decay while approaching the zero crossing during the decay period.

FIG. 10

is a diagrammatic representation of sampling points during attenuation of a lower frequency tone sine wave and during attenuation of a higher frequency tone sine wave.

FIG. 11

is a system diagram illustrating the digital the tone request logic and sample generation logic of the digital signal processor (DSP) of

FIG. 1

in accordance with a preferred embodiment of the invention.

FIG. 12

, including

FIGS. 12A through 12D

, is a flow diagram of an embodiment of the tone attenuation and decay method of the Table 1 embodiment of the invention.

DESCRIPTION OF THE PREFERRED EMBODIMENTS

The invention will be described with respect to three embodiments, including a pseudo-code representation of the tone attenuation and decay methods (Table 1), a C code implementation (Table 2) and a DSP code implementation (Table 3). Generally, the preferred embodiment is that of Table 3. However, for purposes of clarification of various concepts and to illustrate equivalent structures and methods, the embodiments of Tables 1 and 2 are presented.

Glossary and Abbreviations

m

AMPLITUDE CONTROL, aka AMPLITUDE MULTIPLIER.

See ATTENUATION INDEX

α(i)

ANGLE at this sample i

ΔT

ANGLE INCREMENT

142

ATTENUATION PERIOD

248

ATTENUATION INDEX, used to calculate initial

value for AMPLITUDE CONTROL m

β

BANK

2π

CYCLE (sine wave from 0 to 2π)

190

DECAY

DECAY DISTANCE, an approximation of y(i) as a

condition to enter decay

144

DECAY PERIOD

ΔT

DELTA T: angle increment (called “note” in

DSP implementation, and “angleinc” in C code

implementation. These implementations are in

different units. C code is in natural, or

mathematical units, and the DSP code is done

in computationally efficient units.)

Δt

DELTA t: time interval

DAC

DIGITAL TO ANALOG CONVERTER

f

FREQUENCY

y(i)

OUTPUT value of ith sample (digital amplitude

value presented to DAC by DSP)

PCM

PULSE CODED MODULATION

π

Pi = 3.14159 . . .

q

QUADRANT

a,b,c

QUADRATIC COEFFICIENTS

r

SAMPLING RATE (see, SAMPLING INDEX)

r

SAMPLING FREQUENCY (see, SAMPLING INDEX)

r

SAMPLING INDEX

sin

SINE

STEP CONTROL (“dampadd”)

STEP LIMIT (“dampstep” in C code, “atndcay”

in DSP code), a step size related to sampling

frequency

194

STOP

242

TONE INDEX

140

TONE PERIOD

150 &c

ZERO CROSSING (occurs at 0, π, and 2π)

184

ZERO APPROACH POINT

152

ZERO PASSING ZONE

In accordance with the preferred embodiments of the invention, a memory space and processing cycles efficient method and means is provided for computing pulse coded modulation (PCM) values that represent a set of predefined tones. Specifically, digital to analog converter (DAC) inputs are created by a digital signal processor (DSP) that will produce in the DAC an analog electrical signal output to cause one of a set of tones to be produced when applied to an appropriate audio amplifier and speaker.

Attenuation is performed by: (1) reducing the maximum amplitude of the output; (2) reducing the size of the step between two adjacent outputs; and (3) exponentially decaying from the sine to zero. These attenuation actions are applied at certain points in the sine. The amplitude is adjusted when the angle changes quadrant. The step size between two outputs is reduced in a portion of the first and third quadrants, when the sine is moving away from zero. A decision to continue the sine or switch to exponential decay is made in the second and fourth quadrants when the sine is moving toward zero, where the switch may also occur. A continuous function is maintained and, except when the sine value is crossing zero, a continuous first derivative of the function is also maintained. An abrupt but limited change in amplitude occurring when the sine crosses zero does not create objectionable noise.

Referring to

FIG. 1

, a tone generation and attenuation system in accordance with the invention is implemented within digital signal processor (DSP)

102

. DSP

102

receives inputs on line

241

from host processor

100

and on line

135

from phase locked loop (PLL) logic

101

, and selectively on line

247

from audio stream

104

. The output of DSP

102

is fed to digital to analog converter (DAC)

106

, the output of which is fed to amplifier

108

and thence to speaker

118

. PLL logic

101

receives sample index signal

139

from DSP

102

(the sample index value used to generate sample index signal

139

was provided to DSP

102

by host processor

100

or audio stream

104

), and drives sample clock signal

135

to DSP

102

and an over sampled clock signal

137

to DAC

106

. PLL logic

101

locks at the frequency defined by sample clock signal

139

, and responds with a clock signal on line

135

, with each clock signal pulse

135

representing an interrupt request that DSP generate a sample output on line

145

to DAC

106

.

Referring to

FIG. 2

, DSP

102

generates digital representations of a selected tone at sampling points

141

during tone period

148

, which includes pure tone period

140

(beginning with sample

143

), attenuate period

142

(beginning with sample

149

), decay period

144

(beginning with sample

184

) and stop

146

. Sampling points

141

are generated at a time interval Δt.

Tone Generation

In the preferred embodiments, DSP

102

generates one of 128 tones, sine waves of 128 different frequencies selected from among 31 different durations. The tones are those of an equal tempered chromatic scale, but could be others with different constants.

Tones are generated using a digital signal processor (DSP)

102

. These processors

102

usually have limited function, providing only fixed point and multiply operations, but not divide operations. The method and system of the invention are particularly useful where few cycles are available in DSP

102

for tone generation.

Referring to

FIG. 3

, PCM data is a digital representation of an analog signal created by sampling the digital value of the signal at fixed time intervals Δt, or by generating digital values representative of the analog signal. In this embodiment, analog sine wave output

147

a

provided from DAC

106

to amplifier

108

on line

147

is generated by smoothing digital values

145

a

,

145

b

received on line

145

from DSP

102

. DAC

106

uses over sampled clock signal

137

to smooth clocked digital signal values received on line

145

from DSP

102

. Typically, responsive to over sampled clock signal

137

and by way of Fourier analysis, DAC

106

does a curve fit to digital values

145

a

,

145

b

sequentially received at rate Δt, such as times

135

a

,

135

b

, respectively, on line

145

to thereby project future points which accumulate in time to define the analog tone signal curve

147

a

, which signal

147

a

is fed on line

147

to amplifier

108

and thence to speaker

118

.

Representative DACs, useful in connection with the DSP of the present invention is the 16 Bit Audio DAC by Crystal (Cirrus Logic), P/N CS4328, and equivalents, such as P/N CS4331 and CS4327, which are 18 and 20 bit Audio DACs, respectively.

Because a tone is a pure sine wave

147

a

, sampling a tone at a fixed time interval Δt from the last sample is the same as calculating the sine of an angle α(i) at a fixed angle increment ΔT from the angle α(i−1) of the last sample. In accordance with the invention, a value representation is provided for making tone generation simple (that is, efficient in processing cycles and memory space) when tone generation is decomposed into two processes: (1) a process for generating a sequence of angles with the appropriate increment ΔT between each adjacent pair of angles; and (2) a process for computing the sine of the angle. Delta T is accumulated to form an angle of which the sine will be calculated to generate a digital tone sample. “Angle” refers to the accumulated delta T's from the beginning of the tone to this, the ith, sample, which is equal to (i)*ΔT.

A user, such as host processor

100

, specifies a tone by providing to DSP

102

a tone index, which is an integer in a range, such as the range 0 to 127 selected for the embodiments described herein. In the equal tempered chromatic scale, the frequency f(i) of note N(i) is

f

(

i

)=2**(1/12)*

f

(

i

−1), (1)

where f(i−1)is the frequency of note N(i−1). Thus, the frequency f(i) of note N(i) is also 2*f(i−12) of note N(i−12) and 1/2*f(i+12) of note N(i+12).

Referring to

FIG. 4

, a table of tones for each of plurality of sampling frequencies is illustrated. In accordance with the preferred embodiment (Table 3) of the invention, user processor

100

or audio stream

104

may specify a sampling rate

240

. In the case of audio stream

104

, the sampling rate is determined by the sampling rate of the audio stream. Alternatively, sampling rate

240

may be determined by prior or current material being played or, as in the embodiment of Table 2, a fixed sampling rate may be hard coded. The increment ΔT′ between angles may be computed as

Δ

T

′=2

*Π*f/r

(2)

which is an increment in radians, and where f is the frequency, r is the sampling rate or frequency, and Π=3.14159 . . . Also,

Δ

T

=65,536

*ΔT

′/2Π (3)

such that one cycle is represented by 65,536 units. In the preferred embodiments described herein, fractional units are carried in 16 bits for high frequency precision.

For a limited number of sampling rates, table

252

provides for each sampling rate

292

,

294

,

296

a list of the angle increments ΔT for the highest frequency of each of the twelve possible note tones

290

. The angle increments ΔT for all lower frequency notes are computed by shifting the highest note increment from table

252

right once for each twelve units (octave) by which the tone index of the highest note and the tone index of the selected note differ.

In the preferred embodiment (Table 3), the six sampling rates accommodated are 16 KHz, 22.05 KHz, 24 KHz, 32 KHz, 44.1 KHz, and 48 KHz (where KHz means kilohertz.) These require three tables

292

,

294

and

296

. The angle increments for the lowest three sampling rates (16 KHz, 22.05 KHz and 24 KHz) are computed by doubling the increment for the higher rate (32 KHz, 44.1 KHz, and 48 KHz, respectively).

ΔT values in table

252

are computed with reference to the American Standard pitch of the equal tempered chromatic scale at A

6

=440 cycles per second (A

6

represents tone A in the sixth octave of the scale).

Referring further to

FIG. 4

, by way of illustration of sampling frequency table

252

, at a sampling frequency of 44.1 KHz, the angle increment ΔT for note

290

tone A in the eleventh octave is ‘51bb.f72d’ (hex, but with a binary decimal separating the two 16 binary bit half words, such that the first half word, herein ‘51bb.’, is equal to or greater than zero, and the second half word, herein ‘.f72d’, is equal to or less than zero). Therefore, to get to A in the sixth octave, this value is shifted right by five binary bits. (In table

252

, the values shown for tones C through G# are in the 10th octave, and A through B are in the 11th octave.) The angle increment for A in the eleventh octave converts to binary:

0101 0001 1011 1011 1111 0111 0010 1101 (

4

)

By shifting five positions to the right, the angle increment for A in the sixth octave is:

0000 0010 1000 1101 1101 1111 1011 1001 (5)

In hex, this is

028

d.dfb

9 (6)

where 028d is greater than 1 and dfb9 is less than 1 due to the binary point.

For a full cycle for A of 440 cycles per second, where a cycle means 65,536 units, the computed ΔT times 44100 samples per second gives a value of 28,835,840 units/second, where, as previously stated, 65,536 units are equivalent to 2Π, or a single sine cycle. Thus,

Δ

T

=028

d.dfb

9 (hex)=653.87392 (decimal) (7)

units per sample.

The total number of units per second is, therefore, 653.87392 times 44,100 equals 28,835,840 (decimal). In accordance with the units implemented in the preferred embodiments of the invention, 65,536 units represent 2Π of angular increment, and the number of cycles per second represented by one second of angular increments as calculated above is 440 (which is the frequency of note A

6

.)

A 31 bit counter, with a binary point in the middle, counts to a largest value of 65,535.999999 . . . (decimal), which means that the counter wraps 440 times per second for A

6

. Similarly, a sin wave

2

Π wraps 440 times in radians for A

6

.

As will be described hereafter in connection with

FIG. 6

, the binary representation of ΔT is accumulated to form a 32 bit value which is α(i), the value in register

271

.

Referring to

FIG. 5

, the relationship between angular velocity ΔT and the sine value for sample (i) is illustrated. For a given sample (i), the angle α(i) is:

α(

i

)=α(

i

−1)+Δ

T

, or (8)

when α(0)=0, then α(

i

)=

i*ΔT

(9)

and the sine value at angle α(i) is represented by value

403

and that for angle α(i−1) by value

402

.

Given an angle, the sine of that angle can be computed with reasonable accuracy from a piecewise continuous curve fitted to the true sine values. If a linear fit is used, more points and somewhat less computation are required. A quadratic fit requires fewer points for the same accuracy and one more add and one more multiply. A cubic fit requires still fewer points for the same accuracy, but is more computationally complex. Any of these can be made to operate to a reasonable accuracy specification. In the preferred embodiment (Table 3) of the invention, the quadratic fit is used and performed with some intermediate shifts to preserve accuracy. In the embodiment of Table 2 C code, the sine is directly calculated.

Referring to

FIG. 6

, the manner in which an angle value is used to select the sine and compute the value of the quadratic is illustrated. This specific embodiment relates to the DSP version set forth in Table 3 at lines

250

through

265

. In this preferred embodiment, increment angle logic

270

provides an output signal

271

comprising two sixteen bit half words

287

and

288

, including sign bit

285

and index bits

286

. Signal

271

is an angle that represent the accumulation of delta T's (ΔT) through the current sample. Compute sine

272

calculates the sine of the angle at the current sample in accordance with the following:

sin(

i

)=(((

a*x

)/2)+

b

)*

x

)+

c

(10)

where a, b and c are values (in hex) selected from table

238

at the row selected by index value

286

and x is the value

289

selected from bits

5

through

19

of signal

287

, with bit position

4

set to zero. The resulting sine value is multiplied by an amplitude (at line

266

of Table 3), rounded and multiplied by the sign of the angle to get the correct quadrant. The result is the output tone, if in the tone period

140

(an not yet executing attenuation). At lines

273

and

274

(Table 3) the code checks if tone period

140

has completed and then branches to an exit routine to wait for the next interrupt on line

135

(FIG.

11

). (In DSP code, the instruction after a branch is always executed.) Referring to

FIG. 7

, a table of tone indexes

242

values

0

through

127

correlated to octave

0

through

10

and notes C (octave

0

) through G (octave

10

) is illustrated.

An important efficiency of the invention is in value representation. In fixed-point arithmetic only values in the range −2**n to 2**n−1 can be represented, where n is the register width in bits. If an add operation would result in a value outside of this range, the result is that value minus 2**n (which is a modulo calculation). Sines of angles have this same characteristic. That is, sin(a) =sin(α−2*Π). Thus, by making 2**n =2*Π, all angles α naturally remain in the range 0≦α<2*Π. Since the sine is represented by a piecewise fit, the values of the sine at the required number of points within the fit range can be computed, and the fit done using the mapped angle values. By choosing fit intervals that are a power of 2 in width, mask and shift operations are sufficient to identify the interval, the coefficients to use, and the value upon which to perform the calculation.

For example, with a 32 bit data width and 16 intervals from 0 to Π, the angle α is interpreted as:

bit 0:

sign of the result.

bits 1-4:

index of the fit interval.

bits 5-19:

sine value, x below.

The approximate sine is calculated as:

abs(sin(α))=(

a*x+b

)*

x+c

(11)

where a, b, and c are values obtained from the sine table

238

. The sign of the above result can then be changed, if necessary (3rd or 4th quadrant), based upon the bit

0

value. As implemented in the DSP code embodiment of the invention (Table 3), in order to optimize machine components and cycles, the quadratic calculation of the approximate sine is:

abs(sin(α))=(((

a*x

)/2)+

b

)*

x

)+

c,

(12)

with rounding occurring after (a*x), as implemented at lines

261

through

265

of the DSP code implementation of Table 3.

In the preferred DSP code (Table 3) embodiment of the invention, the table of notes per sampling rate and the table of coefficients of the piecewise fit to the sine are computed and stored either in a ROM or in initialized values of a RAM, thus avoiding code for their calculation in the DSP. The table of notes is calculated as:

((2

**n

)*

f

)/

r

(13)

where n is the data width, f is the frequency, and r is the sampling rate. In the table, the low order four hex digits are fractional.

During pure tone period

140

, a tone output (PCM data) is generated by DSP

102

by calculating the sine of an angle a which is being increased at a constant rate. Each output signal y(i) is computed by adding an increment ΔT to the angle α(i−1), calculating the sine of the angle α(i), then scaling the resulting value to a required range by multiplying by an amplitude multiplier m where m≦1, the initial value of m is determined by the attenuation value index

248

, and the attenuation value index

248

is, for example, a three bit binary number selecting one of eight reduction factors.

Thus, during pure tone period

140

, the sample value y(i) of the tone generated for i′th sample

141

is given by equation (15), as follows:

y

(

i

)=

m

*sin(α(

i

)) (15)

where m is derived from the attenuation value index

248

, α(i) is the angle, which is i*ΔT, and y is the output value

278

.

As will next be described, attenuation of the tone following pure tone period

140

follows the same approach, but the amplitude multiplier m of the output signal is modified, and the output signal is further modified to achieve attenuation in the required time

142

,

144

.

Tone Attenuation

In general, in accordance with the invention, tone attenuation during attenuate period

142

includes attenuation at zero crossings and attenuation during zero passing zones. This is followed by a decay period

144

, followed by stop

146

. The attenuation during the attenuate period, particularly within zero passing zones, results in a moderately disturbed but continuous sine of decreasing amplitude.

During tone attenuate period

142

, the amplitude m of the tone is reduced at each zero crossing in accordance with equation (16), as follows:

m =z*m

where (16)

where z is the attenuation adjustment value for zero crossings, and is set heuristically at some value between approximately ½ and ¾. This adjustment of the attenuation multiplier m is performed prior to the calculation of the first sample following the zero crossing. Thus, ignoring further attenuation adjustments, the next half wave would be of amplitude z*m, and the jth half wave in the attenuation period would have amplitude m*z**j.

Referring to

FIG. 8

, also during attenuate period

142

, the amplitude y(i) of the tone generated is attenuated following each zero crossing (in the zero passing zone, or interval

152

). Curve

160

represents y′(i), which equals

y

′(

i

)=

m

*sin (α(

i

)). (17)

The actual y(i), or curve

164

in the zero passing zone

152

, is calculated with reference to y′ (i) as follows. Let i(

0

) represent the index of the first sample in the zero passing zone. For the first sample

153

in the zero passing zone, β of i(0)=0, and y(i(0))=y′ (i(0)). With respect to the second sample

155

and subsequent samples in zero passing zone

152

, bank is derived as follows:

β(

i

)=β(

i

−1)+((

y

′(

i

)−

y

′(

i

−1))−(

d

**(

i−i

(0))*(

y

′(

i

)−(

y

′(

i

−1)) (18)

where d is “dampadd” in C code Table 2, y′(i) is “temp”, y′(i−1) is “temp

1

” and d**(i−i(0)) is “damp”.

The output y(i) is calculated as follows:

y

(

i

)=

y

′(

i

)−β(

i

) (19)

Bank β(i(0)+1) is represented by value

159

. y(i) is curve

164

, and y′(i) is curve

160

, in intervals

152

and

154

.

During interval

152

, β(i) is modified according to equation (

18

). In interval

154

, β(i)=(i−1), or in other words, the bank is not modified.

At the boundary between intervals

154

and

156

,

m=m−β.

(20)

In the second and fourth quadrants (interval

156

, etc.) the output value y(i) is calculated as follows:

y

(

i

)=

m

*sin (α(

i

)). (21)

The m in the first quadrant is

162

. The m in the second quadrant is amplitude

166

, which equals amplitude

162

minus the bank β(i)

172

,

174

throughout interval

154

, which is a constant value. Value

170

represents the β(i+k), where i+k is some sample time following i(0)+1 in interval

152

. Through point

153

, curves

160

and

164

coincide.

While the above discussion of attenuation refers to the first and second quadrants of the sine wave, the same principles apply in the third and fourth quadrants.

In the embodiments of Tables 2 and 3, the conclusion of the attenuation period

142

is determined differently. In C code Table 2, the iteration that produces the set of output values is part of the code. For simplicity, the pure tone period

140

and the attenuation period

142

are a single iteration starting at Table 2 line

86

characterized by the computation of a sine. The decay period is a separate iteration starting at line

136

.

In DSP code Table 3, the iteration is external, driven by the PLL sample interrupts represented by line

135

. The entry point for sample generation is at line

209

. The test for decay period occurs at

214

and the branch to decay code occurs at line

215

. What was two separate iterations in the C code Table 2, is two separate paths in the DSP implementation.

Referring to

FIG. 9

, beginning of the decay period

144

at point

184

is recognized when the following three conditions are met:

First, the angle is within the interval

157.50°≦α≦180° or 337.5≦α<180° (22)

Second, the damp s is less than dampstep:

s

≦dampstep (23)

where dampstep is a sampling rate related value, and is a bound on the step size that assures that the velocity of the speaker is not too high as decay period is entered. As a speaker

118

velocity related value, it is related to sampling rate (lower for high sampling rates, and higher for low sampling rates). In the DSP code Table 3, this value for dampstep is calculated at line

193

and is a constant in the C code which is only valid for sampling frequency 44.1 Khz.

Third, the y(i) at point

184

satisfies the following inequality:

6

*s≦|y

(

i

)|≦8

*s

(24)

where

s=abs

(

y

(

i

)−

y

(

i

−1)). (25)

Thus, a value for y(i) is selected to start decay which allows a smooth transition into the decay period from the attenuation period. Thus, the transition to decay is that of a substantially continuous function. This determination is made in similar ways in the C code and DSP code embodiments. In the C code, this calculation is determined as y(i) is less than ⅜ amplitude. In the DSP code, the quadratic is 13 to 15, which is related to the angle (the last ⅜ths of the second or fourth quadrant).

Referring to

FIG. 9

, exponential decay period

144

generates an exponential decay from the point

184

on sine wave

176

to zero at stop sample point

194

along path

190

. Point

184

, on sine wave

176

of amplitude

178

in attenuation period

142

, is the zero approach point, the first point that meets the three conditions above at equations 22-25 for starting decay.

y

(

i

)=⅞*(

y

(

i

−1)) (26)

where ⅞ is a heuristic value for the decay constant. In alternative embodiments, the decay entry conditions and this constant would need to change together in a manner to achieve a smooth transition from the sine wave

176

to the decay curve

190

.

At stop

146

, which occurs with the sample immediately following the last sample in decay period

144

before the zero crossing,

y

(

i

)=0. (27)

This decay process may be skipped if the attenuation produces two sequential zero value samples for y(i).

Referring to

FIG. 10

, two tones in attenuation are illustrated: one tone

220

of relatively high frequency and the other tone

200

of relatively low frequency. A few illustrative sample points

206

,

208

,

210

,

212

,

214

,

216

are illustrated along sine wave

200

and points

224

,

226

,

230

and

232

along sine wave

220

. In attenuate period

142

, high frequency tone

220

will have (1) many zero crossings

222

,

228

, . . . ; (2) few consecutive outputs in the first half of the first or third quadrants (no such consecutive outputs are shown in

FIG. 10

for tone

220

); and (3) a large step size versus output value when tested in the second and fourth quadrants. As a consequence, attenuation of a high frequency tone will be accomplished largely by zero crossing attenuation (factor z, referred to as amplitude control in Table 1). The step control, equivalent to dampadd in the C code, will have no or minor effect, because very few sample points occur in the zero passing zones of the first and third quadrants (shown in

FIG. 10

are only sample points

224

and

230

which appear to occur in this zone for tone

220

). Inasmuch as successive output values will not meet the requirements to enter exponential decay, attenuation at zero crossings

222

,

228

. . . is relied upon to cause the output to go to zero. No exponential decay will occur, and stop will be recognized by two consecutive zero values on the output. (If the tone frequency is close to the sample frequency, two successive zero sample values may occur at zero crossings, but this would be a contradiction of generally accepted tone frequency sampling frequency relationships which require that the tone frequency be something less than the sampling frequency. For instance, in accordance with the Nyquist principle, the highest frequency that can be reasonably produced at a given sampling rate is the sampling rate divided by 2.2.)

Referring further to

FIG. 10

, in the attenuation period, a low frequency tone

200

will have (1) very few zero crossings

202

,

204

. . . ; (

2

) many consecutive outputs

206

,

212

in the first half of the first or third quadrants; and (

3

) a small step size

180

(

FIG. 9

) versus output value when tested in the second and fourth quadrants, such as at samples

216

. As a consequence, the attenuation of the low frequency tone

200

will be much more effected by the step control (“dampadd”) and the low frequency tone will meet the requirements for exponential decay

217

to be applied.

Tones of intermediate frequency are attenuated with a combination of the actions. Thus, if tones of high and low frequency attenuate in the required time, tones of intermediate frequency will also attenuate in the required time.

From the pseudo code of Table 1, it is apparent that none of the control decisions nor the value modifications require more than a few instructions to implement. Also, the number of controls and the number of stored values is also small. This fulfills the objective that the solution be small in both code and data space.

Referring to

FIG. 11

, which represents the common elements of the three embodiments of Tables 1, 2 and 3 of the system of the invention, digital signal processor

102

of

FIG. 1

includes tone request logic

235

and sample generation logic

237

.

Host processor

100

inputs to DSP

102

, represented by line

241

, include sampling index

240

, tone index

242

, duration index

246

, and attenuation value index

248

. Alternatively, sampling index

240

may be loaded from audio stream

104

. As represented by lines

139

and

243

, sampling index

240

is an input to PLL

101

, shift value

250

, tone table

252

and sample count logic

260

. As represented by line

245

, tone index

242

is an input to shift value

250

and tone table

252

. As represented by line

249

, duration index

246

is an input to sample count

260

. As represented by line

257

, the value m initialized by attenuation value index

248

is an input to adjust for attenuation logic

274

. As is represented by line

253

, the output of tone table

252

is an input to tone value

254

, the output of which is an input represented by line

255

to delta T (ΔT) logic

256

. As represented by line

251

, the other input to delta T

256

is the output of shift value

250

.

Sample count

260

is decremented under control of decrement logic

262

. Sample count

260

is initialized by sampling index

240

and duration index

246

. Sample

260

is decremented by decrement logic

262

for each sample output produced and to define three states: decremented during tone state

264

, which provides a true signal represented by line

265

to increment angle

270

during tone period

140

; held at one during attenuate state

266

, which provides a true signal represented by line

267

to increment angle

270

during attenuation period

142

; held at one during decay state

144

, which provides a true signal represented by line

269

to ⅞ output logic

276

during decay period

144

; and set to zero on stop state. All interrupts

135

are serviced until sample count

260

is set to zero.

As is represented by line

271

, the incremented angle, which is an output of increment angle logic

270

, is an input to compute sine logic

272

, the output of which, as is represented by line

273

, is an input to adjust for attenuation logic

274

. As is represented by line

275

, the output of adjust for attenuation logic

274

is fed to output latch

278

and on line

145

to DAC

106

. As is represented by line

279

, the output of output register

278

is fed to

7

/

8

output logic

276

.

In operation, tone request logic

235

receives a tone request from user

100

,

104

and prepares to generate a digital representation of the tone by establishing the angle increment value ΔT

256

and generating a request to PLL

101

for sampling interrupts at the frequency specified by sampling index

240

. Alternatively, the PLL

101

may be running at a given sampling index in response to an audio stream. Responsive to sample interrupts from PLL

101

on line

135

, sample generation logic

237

generates digital representations of the tone signal throughout tone period

148

to DAC

106

.

Attenuation value index

248

represents a tone sound level from which factor m is derived, which factor m is the factor used to adjust a maximum possible amplitude to the amplitude desired by the user during tone period

140

, and is also the initial value for the amplitude at the beginning

149

of tone attenuation period

142

. Index

248

is an index to the initial value of a multiplier on the sine required to take a sine value from the range −1 to +1 into the range −32768 to +32767. (In the preferred embodiment, this entire range is not covered, but is scaled down by about 3 db to keep away from computational edges which prevent calculation of the sine due to changes in sign caused by register overflows.) This index

248

and will be set to “0” for the loudest sound. In the C code implementation of Table 2, the index

248

is not included, but rather the value m is hard coded. In the DSP code implementation of Table 3 (lines

123

through

130

), attenuation value index

248

is interpreted as an index into a table of multiplier values representing approximately 3 db increments.

For these embodiments, DAC

106

accepts output values in the range 32,767 to −32,768. However, the system is not limited to 16 bit output, and could be made to accommodate larger output value ranges.

During tone period

140

and also at the beginning of attenuation period

142

, adjust for attenuation logic

274

takes the product of computed sine

272

on line

273

and of the attenuation value on line

257

, and provides output

278

.

Increment angle logic

270

calculates the angle

271

for sample i as the sum of ΔT

256

and angle

271

for the previous sample i−1. ΔT

256

is an increment, a constant angular increment that is used to create sine value

273

.

Tone index

242

from processor

100

is used to derive a shift value

250

and to access tone table

252

to derive a tone value

254

. Shift value

251

and tone value

254

are used to derive ΔT

256

. It is a characteristic of and advantage of the Table 3 DSP code implementation that most of the computational complexity is included in deriving the table of values of ΔT

256

, which may now be determined by a selection and a single shift operations.

Referring further to

FIGS. 4 and 7

, tone index

242

is a value from 0 for C in octave

0

to

127

for G in octave

11

. Tone index

242

is taken modulo

12

to give a number

291

in the range 0 to 11, which maps into notes

290

, C through B, in tone table

252

. Tone index

242

is also divided by 12 to give a value which is subtracted from 10 to give shift value

250

. Tone value

254

is shifted right by shift value

250

to obtain ΔT

256

. The value ΔT is also represented in

FIG. 5

, where the angle of this sample i is related to the angle of the previous sample i−1 by the value ΔT:

α(

i

)=α(

i

−1)+Δ

T.

(28)

In response to an interrupt on line

135

, control is transferred to sample generation logic

237

for the generating a single sample in response to the interrupt, which occur at the frequency (samples per second) specified in sampling index

240

. In response to the interrupt, sample generation logic

237

decrements sample count

260

, increments angle

271

, and computes sine

273

. Based on sample count

260

, a decision is made to change states

264

,

266

and

268

to tone period

140

, attenuate period

142

, or decay period

144

, respectively.

In the DSP code implementation, when in decay state and output signal

279

equals zero, then sample count

260

is set to zero. When in attenuate state, two consecutive outputs are zero, then the sample count is also set to zero. During pure tone period

140

(tone state

264

is true), output

278

is driven by adjust for attenuation logic

274

. At any particular time, angle

271

is equal to i * ΔT, where i is the integer label

141

, which increases with each sample

141

starting with

0

at the beginning

143

of tone period

140

. With each interrupt

135

, ΔT is added to the angle for the previous interrupt

141

(i−1) to get the angle for the current interrupt

141

(i). In C code Table 2, at line

65

, “angleinc” is the same as ΔT, except it is in radians. ΔT in the DSP code is “note” in the 65K=2Π units.

Referring further to

FIG. 11

in connection with

FIG. 2

, at the beginning of pure tone period

140

, sample count

260

is set to duration index

246

times a value selected by sampling index

240

. In the preferred embodiment, the initial sample count

260

value is the number of equal time value durations, expressed in number of samples, set by host processor

100

in duration index

246

, minus an average number of attenuation and decay samples, so as to initialize sample count

260

to the number of samples required during pure tone period

140

. During pure tone period

140

, while sample count

260

is counting down to zero, attenuation value m initialized by attenuation value index

248

, drives adjust for attenuation logic

274

. When sample count

260

has decremented to zero, attenuation period

142

is entered, sample count

260

is no longer decremented, and adjust for attenuation

274

is driven by m and bank value β

172

(as further described with respect to FIG.

8

). Values m and β are modified by selected angles during attenuation period.

Attenuate period

142

is recognized, and attenuate state

266

made true, by sample count

260

being 1 prior to decrementing.

Referring to

FIG. 12

, including

FIGS. 12A through 12D

, the method of the invention set forth in the embodiment of Table 1, is illustrated. Selected process steps

300

-

372

in

FIG. 12

are annotated to the code of Table 1. In step

300

, a request for a tone is received from processor

100

and the request parameters loaded into sampling index

240

, tone index

242

, duration index

246

and attenuation value index

248

. In step

302

, delta T

256

is derived as heretofore explained. The WHILE of line

1

of Table 1 is a representation of the repeated sample interrupts from PLL

101

. Processing then continues as set forth in Table 1.

In Table 1, a pseudo-code representation of the tone attenuation and decay methods of the invention is set forth. In this representation of the method of the invention, AMPLITUDE CONTROL is the fraction by which to reduce the amplitude on zero crossings; INITIAL STEP CONTROL is the fraction by which to reduce the step control; DECAY DISTANCE is the step size multiplier that characterizes the decay rate; and DECAY RATE is the fraction by which to multiply the last output to obtain the current output while in exponential decay. The decay rate and decay distance are related as follows:

r

=decay rate; (29)

decay distance=1/(1

−r

)=1

+r+r

**2

+r

**3 (30)

For example, if decay rate is ⅞ then decay distance is 1/(⅛) or 8.

TABLE 1

Pseudo-Code Representation

306

UNTIL ((CURRENT SINE IS EQUAL TO ZERO) AND (LAST

SINE IS EQUAL TO ZERO) OR (FLAG));

LAST SINE = SINE;

312

SINE = AMPLITUDE * SIN(ANGLE);

314

OUTPUT = SINE;

316

IF SINE JUST CROSSED ZERO,

THEN DO:

318

AMPLITUDE = AMPLITUDE * AMPLITUDE

CONTROL;

320

OUTPUT = OUTPUT * AMPLITUDE CONTROL;

322

STEP CONTROL = INITIAL STEP CONTROL;

END;

326

ELSE IF SINE IS IN THE FIRST HALF OF THE FIRST OR

THIRD QUADRANT,

THEN DO:

328

STEP = CURRENT SINE − LAST SINE;

330

BANK = BANK + STEP − STEP CONTROL * STEP;

332

STEP CONTROL = STEP CONTROL * INITIAL STEP

CONTROL;

END;

336

ELSE IF REDUCTION IS NOT ZERO,

THEN DO:

338

AMPLITUDE = AMPLITUDE − BANK;

340

OUTPUT = OUTPUT − REDUCTION;

342

BANK = 0;

END;

346

ELSE IF ANGLE IS IN THE LAST THIRD OF THE

SECOND OR FOURTH QUADRANT,

348

THEN IF ABSOLUTE STEP < STEP LIMIT,

350

THEN IF DECAY DISTANCE * STEP =

OUTPUT,

352

THEN FLAG = TRUE;

360

OUTPUT = OUTPUT − BANK;

362

WRITE OUTPUT;

364

ANGLE = ANGLE + ANGLE INCREMENT

END;

366

WHILE (LAST OUTPUT IS NOT ZERO)

368

OUTPUT = DECAY RATE * OUTPUT;

370

WRITE OUTPUT;

372

END;

TABLE 2

Beep Generation I (C-Code)

#include <stdio.h>

#include <string.h>

#include <math.h>

void main()

{

int i, octave, note;

int index;

int newdelta;

int ampi;

int bank;

int points;

int diff;

int j;

int tlim;

int anglep, angleint;

int short temp;

int short temp1;

int short out;

double cycle;

int duration = 7;

int dursamp;

double ffreq[12];

/* computed frequency of highest

double angle;

tones */

double angleinc;

double damp;

double samp = 44110.0;

/* sampling rate */

int dampstep = 110;

double dampinit = .625;

/* 5000 / 8000 */

double dampadd = .9863281;

/* 7E40 / 8000 */

int zize;

int flg;

/* angle increments for other sampling rates obtained

/* by modifying variable “samp” above */

int notes[12];

/* computed angle increments */

double PI;

char out_name[64] = (“pcmout.pcm”);

FILE *fopen(), *pcmout;

/* compute highest frequency tone increments from an “A” =

440 * 2**5 */

angleinc = pow(2, (double)1/12);

ffreq[9] = 14080.0;

ffreq[10] = ffreq[9] * angleinc;

ffreq[11] = ffreq[10] * angleinc;

for (i=8; i>=0;i--) ffreq[i] =

ffreq[i + 1] / angleinc;

for (i=0;i<12;i++) notes[i] = ffreq[i] *

(65536.0 * 65536.0 / samp) + .5;

PI = 3.14159265358979;

pcmout = fopen*(out_name,“wb”);

j = 46;

/* can change to generate other tones */

ampi = 0x00005a82;

/* .707107 . . .

index = j;

octave = index / 12;

/* convert tone index into note and

/*octave */

note = index % 12;

/* calculate angle increment for the tone in double and

/* int.

/* calculate tone cycles per second for information and

/* reference. */

angleinc = notes[note] / 65536.0;

for (i=10;i>octave;i--) angleinc / = 2;

cycle = (samp * angleinc)/(65536.0);

/* for reference */

anglep = angleinc * 65536.0;

/* int value of angle in DSP */

/* solution, 65536 = 2* PI */

angleinc = angleinc * 2 * PI / 65536;

/* generate cycle Hz tone at samp Hz sampling

*/

/* freguency for duration / 10 sec

*/

points = 0;

temp1 = 0;

angle = 0;

dursamp = (duration * samp )/10.0;

for (i=0; i<dursamp; i ++)

{

templ = temp;

temp = ampi * sin((angle));

angle = angle + angleinc;

angleint = angleint + anglep;

if (angle > 2*PI) angle = angle − 2*PI;

zize=fwrite(&temp,sizeof(temp),1,pcmout);

points++;

}

/* attenuate tone */

bank = 0;

damp = dampadd;

/* start damping if in 1

st

or */ /* 3

rd

quadrant */

flg = 0;

while (temp!=0 | | temp1!=0)

{

temp1 = temp;

temp = ampi * sin((angle));

/* if crossing zero, change amplitude, adjust */

/* temp, initialize additional damping values */

if (temp / abs(temp) !=templ / abs(temp1))

{

ampi = ampi * dampinit;

temp = temp * dampinit;

damp = dampadd;

}

/* if going away from zero, newdelt=damp*delta */

/* must do compare on angles on high frequency */

/* tones */

else if ((0.0<angle&&angle<PI/2.0) | |

PI < angle&&angle<3.0PI/2.0))

{

newdelta = temp − temp1;

bank = bank+newdelta − (int) (newdelta*damp);

/* if damp > .4) */

if (abs(temp1) < (71*ampi)/100)

damp = dampadd * damp;

}

/* check if just changed direction

*/

/* crossed PI/2 or 3P1/2

*/

else if (bank)

{

ampi = ampi − abs(bank);

temp = temp − bank;

bank = 0

}

/* check if nearing zero, angle nearing

*/

/* zero or PI

*/

else

{

if (abs(temp) < 3*ampi/8

if (abs(temp − temp1) <= dampstep)

if( (8*abs(temp − temp1)>=abs(temp)) &&

(abs (temp)>= 6*abs(temp-temp1))

flg = 2;

}

out = temp − bank;

angleint = angleint + anglep;

angle = angle + angleinc;

if (angle > 2*PI) angle = angle − 2*PI;

zize = fwrite(&out,sizeof(temp),1,pcmout);

points++;

If (flg ==2) break;

}

/* exponential decay to zero */

while (temp1 != 0)

{

temp1 = temp;

temp = (7*temp1)/8;

zize=fwrite(&temp,sizeof(temp),i,pcmout);

points++;

}

/* add some trailing zeros to guarantee a quite moment */

for(i=0;i<1024;i++)

{

zize=fwrite(&temp,sizeof(temp),1,pcmout);

}

}

Referring to Table 3, the DSP assembly language embodiment of the invention is set forth. The DSP code implementation differs from the C code implementation of Table 2 in that in the DSP code a change in sampling frequency during the tone generation period is accommodated without changing the audible tone. The output of DAC

106

will be substantially the same for small changes in sampling frequency. For instance, a change from a sampling frequency of 44.1 KHz to 48 KHz is not detectable by a human. The C code implementation supports only a single sampling rate (44.1 KHz). Also, it computes some of the table values that the DSP code reads. For example, the sine is computed by the system in the C code implementation, rather than by the spline fit table used by the DSP code. A pseudo code representation of the algorithm executed by DSP code is set forth in Table 3 at lines

26

through

86

, and the remainder of the code is generously commented. The DSP code language syntax used in Table 3 is described in “

Mwave Development Tookkit, Assembly Language Reference Manual

”, Intermetrics, Inc, Cambridge, Mass., copyright 1992, 1993.

TABLE 3

============================================================================================

Beep Generation II (DSP Code)

============================================================================================

8

;********************************************************************************************************

9

;* Beep generation code

10

;*

11

;*

Not & subroutine, strictly speaking, since it does not return to the

12

;*

caller. Split off like this to make it easier to move to RAM.

13

;*

14

;* NOTE: Don't need to make the return statement flexible, since if we move

15

;*

this code to RAM, it will already return to the correct spot. If

16

;*

Sample gets moved to RAM, this code is still useable, since

17

;*

nothing really gets done after this routine finishes.

18

;*

19

;* Variables

Description

20

;* ---------

-------------

21

;* nnotes

The angular increment for the comment note in the

22

;*

highest octave.

23

;* dur_mult

The number of samples at 48K times dur_mult / 800 base

24

;*

16 is the number of samples at the actual sampling

25

;*

frequency.

26

;* dur_recp

The number of samples at the actual frequency *

27

;*

dur_recp / 8000 base 16 is the number of samples

28

;*

at 48K. These are approximate.

29

;* atndcay

Maximum step size to switch to decay in 2nd or 4th

30

;*

quadrants.

31

;*

32

;********************************************************************************************************

33

;* Tone Initialization and Play

34

;*

on entry to initializaion, r1 = contents of PCM_CON,

35

;*

encoded duration and attenuation

36

;*

37

;* Tone Initialization

38

;* - Calculate attenuation from attenuation index in PCM_CON.

39

;* - Calculate note and octave from tone index in AUD_CTL.

40

;* - Initialize angle to zero.

41

;* - Save sampling rate.

42

;* - Calculate duration from sampling rate.

43

;* - Sampling rate change reentry point.

44

;* - Calculate note from noteidx.

45

;* - Copy controls that are rate dependent

46

;* - Fall thru into tone pre-process.

47

;*

48

;* Tone Process

49

;*

on entry, wr2 contains duration

50

;* - If final < 1

51

;* -

then out = lastsamp * final

52

;* - Else if sampling rate is not the same,

53

;* -

then

54

;* -

compute new duration

55

;* -

branch to tone initialization reentry point

56

;* -

end

57

;* -

Save sign of angle

58

;* -

Add note to angle

59

;* -

Save sign of updated angle

60

;* -

out = ampi * sine of angle

61

;* -

If duration−1 > 0

62

;* -

then duration = duration − 1

63

;* -

else

64

;* -

out = out − bank

65

;* -

if angle crossed 0 to PI

66

;* -

then

67

;* -

ampi = ampi * zero_atn

68

;* -

out = out * zero_atn

69

;* -

damp = tone_damp

70

;* -

end

71

;* -

else

72

;* -

if angle is in 1st or 3rd quadrant

73

;* -

then

74

;* -

out = out − (out − lastsamp) * (1 − damp)

75

;* -

bank = bank + (out − lastsamp) * (1 − damp)

76

;* -

if in 1st quadrant and angle < 3 PI / 8 OR

77

;* -

in 3rd quadrant and angle < 11 PI / 8

78

;* -

then damp = damp * tone_damp;

79

;* -

end

80

;* -

else

81

;* -

ampi = ampi − abs(bank)

82

;* -

bank = 0

83

;* -

if in 2nd quadrant and angle > 3 PI / 4 OR

84

;* -

in 4nd quadrant and angle > 7 PI / 4

85

;* -

then if abs(out − lastsamp) < tone_step AND

86

;* -

8 * abs(out − lastsamp) >= abs(out)

87

;* -

then final = 7/8;

88

;* -

end

89

;* -

end

90

;* -

end

91

;* - end

92

;* - Store out in left / right output register sources

93

;* - Return

94

;*

95

;*********************************************************************************************************

96

;* Tone constants storage map:

97

;*

98

;* RATE

NOTES

CONVDUR

DURAT

ATNSAMP

ATNSTEP

ATNDCAY

UNUSED

99

;* 00 00000

48

6

2

2

2

2

2

100

;* 01 000000

101

;* 10 000000

102

;*

103

;********************************************************************************************************

104

;*

105

;* Hardware registers

106

;*

107

;*

1

1

1

1

1

1

108

;*

5

4

3

2

1

0

9

8

7

6

5

4

3

2

1

0

109

;*

110

;* PCM_CON

X

X

X

D

D

D

D

D

X

r

r

r

X

A

A

A

111

;* AUD_CTRL

r

T

T

T

T

T

T

T

r

r

r

r

r

r

112

;* FSCR_REG

X

X

X

X

S

S

R

r

r

r

r

r

r

r

113

;*

114

;* r - reserved

X - not relevant

115

;* D - duration

A - attenuation

T - tone index

116

;* S - sampling rate

R - sampling range

117

;*

118

;********************************************************************************************************

119

atn_samps

equ 80

; 320 / 4

120

;PCM.Beep_Req equ ‘1f00’x

; mask to extract duration

121

min_tone

equ 26

; minimum tone index in attenuation

122

; Expects r1 = PCM_CON

123

toni

equ *

124

; calculate tone attenuation from index, clear tone_dcay

125

r6=#7

%r2

126

CDB=r1

r6=r6&r1

127

BIB 0, toni10

; branch LSB attenuation

128

r3=‘4000’x

; n * 6 dB attenuation

129

r3=‘2d41’x

; n * 3 dB attenuation

130

toni10

equ *

131

; 0/1 2/3 4/5 6/7

132

r6=SHR1(r6)

; r6=r6/2, 0, 1, 2, 3

133

r7=#3

134

r6=r6−r7

135

r3=r3*2**r6

136

ampi=r3

137

tone_ash=r6

138

; calculate duration in standard form, 48K sampling rate

139

r7=#PCM.Beep_Req

140

r5=‘004b’x

r1=r1&r7

; r5 = 4800 / 64

141

r7=#atn_samps

r5*r1

; r1 = duration * 256

142

; r7 = attenuation samples / 4

143

wr2=rp

; wr2 = duration * 4

144

wr2=SL(wr2,12)

; r2 = duration / 4

145

r7=#‘00e0’x

r2=r2−r7

; adjusted duration / 4

146

; prepare sampling rate

147

r3=_FSCR_REG

148

r3=SL(r3,-4)

149

r5=#0

r3=r3&r7

150

; clear tone_dcay and angle

151

tone_dcay=r5

152

angle=r5

153

angle+2=r5

154

r1=_AUD_CTRL

; load tone index

155

r1=r1+r1

; isolate note index

156

r1=SL(r1,−9)

157

tone_cur=r1

; save for sampling rate change

158

; sampling rate change reentry point

159

; - r1 = tone_cur

160

; - r2 = duration at 48K divided by 4

161

; - r3 = FSCR_REG shifted right 4 bits

162

toni25

equ *

163

; convert 48K duration to duration for current sampling rate

164

r0=‘00c0’x

; isolate rate selector

165

r0=r0&r3

166

oldfscr=r3

167

r5=dur_mult[r0]

; duration conversion

168

%r4

r5|*|r2

169

wr2=rp

170

wr2=SL(wr2,−13)

171

rS=‘1556’x

172

r5=#48

r1*r5

; tone index * 1/12 to RPH

173

r1=#−10

r4=r4+rpl

; remainder to r4

174

r1=r1+rph

r5|*|r4

; r1 = shift amount,

175

; RPH = 4 * tone index % 12

176

r6=tone_ash

; reload attenuation shift

177

r5=atndcay[r0]

; load decay steps

178

r4=atnstep[r0]

; step attenuation factor

179

r5=r5*2**r6

; shift by attenuation level

180

r0=r0+rph

; note address

181

r7=#‘7fff’x

182

r6=nnotes+2[r0]

; note + 2

183

wr6=SL(wr6,−16)

184

r6=nnotes[r0]

; note

185

CDB=r3

186

BIB 5,toni40

187

wr6=wr6*2**r1

188

wr2=SL(wr2,−1)

; halve duration samples

189

wr6=SL(wr6,1)

; double note increment

190

r5=SL(r5,1)

; double decay step size

191

r4=SL(r4,1)

; double step attn factor

192

toni40

equ *

193

tone_step=r5

; decay step size limit

194

r4=r4&r7

; clear possible sign bit

195

damp=r4

; initialize damp

196

tone_damp=r4

; step damping factor

197

note=r6

198

note+2=r61

199

wr2=%+wr2+1

; guarantee duration {circumflex over ( )}= 0

200

; can remove after testing

201

tone_dur=r2

; save duration

202

tone_dur+2=r21

203

;* Continues on with the rest of Beep code, now that the initial setup

204

;* work has been done!

205

; This is the tone ‘continuation’ point. Come here

206

; if a tone is already playing.

207

; Expects that wr2 has the double word value

208

; for Tone Duration loaded.

209

tone

equ *

210

r3=_FSCR_REG

; r3 = current sampling rate

211

r3=SL(r3,−4)

;

index * 2

212

r5=tone_dcay

213

r7=oldfscr

214

r1=#‘00e0’x

r5

; if decaying, go to decay code

215

bnz tone10

r3=r3&r1

; isolate sampling rate

216

r6=#‘000f’x

r3<>r7

; if sampling rate same

217

bz tone15

r0=r6

;

continue

218

r6=angle

219

;

oldfscr=r3

;

220

; start to recompute set-up

221

; convert current duration to standard form

222

; r1 = old rate, r3 = new rate

223

toni50

equ *

224

r0=#‘00c0’x

225

CDB=r7

r0=r0&r7

; isolate rate

226

BIB 5,toni60

; branch if high rate range

227

wr2=SL (wr2,14)

228

wr2=SL(wr2,1)

; extra shift if low

229

toni60

equ *

230

; note: if 1 < tone_dur < 3

231

;

r2 will be 0

232

; adjusted to 1 after toni25

233

r5=dur_recp[r0]

; conversion reciprocal

234

r5|*|r2

;

235

r2=rpm

;

236

;

237

b toni25

238

r1=tone_cur

239

; exponential decay to zero

240

tone10

equ *

241

r1=lastsamp

; sample = lastsamp * decay

242

r5*r1

; test lastsamp

243

r1=rpm

244

bz tone80

; if sample = 0, tone completed

245

bnz tone91

; else standard exit

246

r1=%+r1+SGN

; add 1 to negative value only

247

tone15

equ *

248

;

r6=angle

; above

249

r61=angle+2

250

r7=SIG r6

; r7 = sign of angle

251

rph=note

; angle = angle + note

252

rpl=note+2

253

wr6=wr6+rp

254

angle=r6

255

angle+2=r61

256

r3=SIG r6

; r3 = sine of angle + note

257

; compute sine of angle

258

; requires that quadratic “c” value be multiplied by 2

259

; separate angle into S Q Q Q Q X X X X X X X X X X X

260

; S - sign, Q - quadratic index, X - value sine = Q(X)

261

wr6=SL(wr6,−11)

; shift

262

;

r0=#‘000f’x

;

above

263

r6=#0

r0=r0&r6

; isolate offset

264

r0=r0+r0

265

wr6=SL(wr6,15)

; isolate X

266

r0=&qa[r0]

; address of quadratic

267

r1=ampi

; amplitude

268

r5=qa-qa(r0)

; a

269

r5=qb-qa(r0)

tnop

r5*r6

; b, a * x

270

r5=r5+rpm+rd

; a * x + b

271

r5=qc-qa(r0)

r5*r6

; c, (a * x + b) * x

272

r5=5+rph+1

; (a * x + b) * x + c + 1

273

r1|*|r5

; multiply * attenuation / 2

274

r5=rpm+rd

275

r5=#1

r3*r5

; multiply by sign

276

r1=rpl

r5*r5

277

; have ampi * sine(angle)

278

; decrement for duration

279

r6=#min_tone

wr3=wr2-rp

mnop

; duration = duration - 1

280

bnz tone90

; if not zero, normal exit

281

tone_dur=r2

282

; attenuation phase

283

r2=tone_cur

284

r4=oldfscr

285

r5=#1

r2<>r6

; if tone_cur >= tone_min

286

bnl tone20

r4=r4+r5

;

then branch, no adjustment

287

r2=r2+r5

; increase tone index

288

; force miscompare without

289

oldfscr=r4

;

changing rate

290

tone_cur=r2

; save updated tone

291

; did sine cross zero in interval?

292

tone20

equ *

293

r3<>r7

; sign of last angle vs current

294

bne tone70

; branch if changed

295

r7=bank

;

296

r6=#16

r1=r1−r7

; sample = sample = bank

297

r4=lastsamp

298

r6=#12

r0<>r6

; compare quadrant

299

bnl tone40

r4=r4−r1

; branch 2nd or 4th quadrant

300

r7

;

temp = (samp − lastsamp)

301

; test bank

302

; first or third quadrant

303

r5=damp

304

r4*r5

; t1 = tamp * damp

305

r4=r4−prm−rd

; temp = temp − t1

306

r1=r1+r4

; samp = samp − temp

307

r7=r7−r4

; bank = bank + temp

308

r0<>r6

; is angle >= 67.5 degrees

309

bnl tone30

r4=r5

; branch yes, r4 = damp

310

bank=r7

311

r5=tone_damp

;

312

r4*r5

; damp damp * dampatn

313

r4=rpm+rd

314

tone30

equ *

315

b tone91

316

damp=r4

317

; second or fourth quadrant

318

tone40

equ *

319

bz tone50

%r2

; if bank {circumflex over ( )}= 0

320

bank=r2

; bank = 0

321

r5=#‘5a82’x

; adjust for full scale

322

r7=|r7|

; |bank|

323

r5*r7

; |bank| * full scale value

324

r3=ampi

325

r3=r3−rpm

; ampi = ampi − bank

326

b tone91

;

327

ampi=r3

328

tone50

equ *

329

r3=#28

r4=|r4|

330

r7=tone_step

;

331

r5=#6

r0<>r3

; test decay start angle

332

; branch if not near end

333

;

of quadrant

334

bl tone91

r4<>r7

; step <> decay step

335

r5*r4

; 6 * |step|

336

; test if 6*|step| <= |sample| <= 8*|step|

337

r4=SL(r4,3)

; 8 * |step|

338

; branch if step is too large

339

bh tone91

r3=|r1|

;

|sample|

340

r4<>r3

; 8 * |step| <> |sample|

341

bl tone91

; branch |sample| > 8*|step|

342

r2=#‘7000’x

343

r3−rpl

; |sample| − 6*|step|

344

bn tone91

; branch |sample| < 6*|step|

345

bnn tone91

; branch in range

346

tone_dcay=r2

; start decaying

347

; sine crossed zero and attenuating

348

tone70

equ *

349

r3=tone_damp

; reinitialize damp

350

damp=r3

351

r3=ampi

352

r5=zero atn

353

r1*r5

; adjust sample value

354

r1=rpm+rd

r3*r5

; adjust attenuation

355

r3=rpm

356

bnz tone91

; no duration update exit

357

ampi=r3

358

; ampi = 0

359

; set duration = 0 − end of tone

360

tone80

equ *

361

wr2=wr2−wr2

; force duration to zero

362

tone90

equ *

363

tone_dur+2=r21

364

tone91

equ *

365

lastsamp=r1

366

_SMP_F0=r1

;

@13A

367

b SMP_EXIT

;* this is the end of Beep. Returns control here.

368

_SMP_F1=r1

;

@13A

============================================================================================

TABLE 4

===========================================================

Quadratics Sine Fit Table

===========================================================

8

;* coefficients - 16 piecewise continuous quadratics fitted to sine of 0 to PI

9

ROM ‘qa’,

‘ff51’,‘fdfa’,‘fcb6’,‘fb94’,‘fa9c’,‘f9da’,‘f954’,‘f90f’

10

ROM ‘’,

‘f90f’,‘f954’,‘f9da’,‘fa9c’,‘fb94’,‘fcb6’,‘fdfa’,‘ff51’

11

ROM ‘qb’,

‘4750’,‘45f1’,‘41e1’,‘3b49’,‘326a’,‘279b’,‘1b46’,‘0de5’

12

ROM ‘‘,

‘fffc’,‘f212’,‘e4b2’,‘d85e’,‘cd90’,‘c4b2’,‘be1c’,‘ba0e’

13

ROM ‘qc’,

‘0000’,‘2351’,‘4546’,‘6492’,‘8000’,‘9683’,‘a73d’,‘b18b’

14

ROM ‘‘,

‘b505’,‘b18b’,‘a73d’,‘9683’,‘8000’,‘6492’,‘4546’,‘2351’

===========================================================

TABLE 5

==============================================================================

Tone Constants Storage Map

==============================================================================

8

;************************************************************************************

9

;* Tone constants storage map:

10

;* RATE

NOTES

dur_mult

dur_recp

atnstep

atndcay

unused

11

;* 00 000000

48

2

2

2

2

8

12

;* 01 000000

13

;* 10 000000

14

;*

15

;* Variables include:

16

;*

nnotes

The angular increment for the comment note in the highest

17

;*

octave.

18

;*

dur_mult

The number of samples 48K times dur_mult / 8000 base 16

19

;

is the number of samples at the actual sampling

20

;

frequency.

21

;****************************************************************************************

22

;* american pitch, A=440 cps

23

;*

first note set is “C”, note is an angle increment in dword

24

;* 44.1K values

25

;*

C

C#

D

D#

26

ROM ‘nnotes ’,

‘3099’,‘76df’,‘337d’,‘45b6’,‘368d’,‘1251’,‘39cb’,‘7a59’

27

;*

E

F

F#

G

28

ROM ‘’,

‘3d3b’,‘4348’,‘40df’,‘5cc9’,‘44ba’,‘e33a’,‘48d1’,‘2253’

29

;*

G#

A

A#

B

30

ROM ‘’,

‘4d25’,‘97f5’,‘51bb’,‘f72d’,‘5698’,‘2b55’,‘5bbe’,‘5b72’

31

ROM ‘dur_mult’,

’759a’

32

ROM ‘dur_recp’,

‘8b52’

33

ROM ‘atnstep’,

‘7ccd’

34

ROM ‘atndcay’,

‘00b0’

35

ROM ‘‘,

‘0000’

36

ROM ‘‘,

‘0000’

37

ROM ‘‘,

‘0000’

38

ROM ‘‘,

‘0000’

39

;*

40

;* 48K values

41

;*

C

C#

D

D#

42

ROM ‘ ’,

‘2ca6’,‘986a’,‘2f4e’,‘4b3f’,‘321e’,‘68d4’,‘3519’,‘5868’

43

;*

E

F

F#

G

44

ROM ‘’,

‘3841’,‘a5d1’,‘3b9a’,‘03a6’,‘3f25’,‘4d91’,‘42e6’,‘8abc’

45

;

G#

A

A#

B

46

ROM ‘’,

‘4Ge0’,‘f069’,‘4b17’,‘e4b1’,‘4fBf’,‘016a’,‘544a’,‘1737’

47

ROM ‘ ’,

‘8000’

48

ROM ‘ ’,

‘8000’

49

ROM ‘ ’,

‘7da3’

50

ROM ‘ ’,

‘0093'

51

ROM ‘’,

‘0000’

52

ROM ‘’,

‘0000’

53

ROM ‘’,

‘0000’

54

ROM ‘’,

‘0000’

55

;*

56

;* 32K values

57

;*

C

C#

D

D#

58

ROM ‘’,

‘42f9’,‘e49f’,‘46f5’,‘70df’,‘4b2d’,‘9d3e’,‘4fa6’,‘049c’

59

;*

E

F

F#

G

60

ROM ‘’,

‘5462’, ‘78b9’,‘5967’,‘0579’,‘5eb7’,‘f459’,‘6459’,‘d01a’

61

;*

G#

A

A#

B

62

ROM ‘’,

‘6a51’,‘689e’,‘70a3’,‘d70a’,‘7756’,‘821e’,‘7e6f’,‘22d3’

63

ROM ‘ ’,

‘5556’

64

ROM ‘ ’,

‘c000’

65

ROM ‘ ’,

‘7b98’

66

ROM ‘ ’,

‘00fc’

67

ROM ‘’,

‘0000’

68

ROM ‘’,

‘0000’

69

ROM ‘’,

‘0000’

70

ROM ‘’,

‘0000’

==============================================================================

ADVANTAGES OF THE INVENTION

It is, therefore, an advantage of the invention that a digital signal processor efficient in a memory space and processing cycles is used to generate and attenuate tones.

It is a further advantage of the invention that a tone is attenuated without creating additional sounds or artifacts at the end of the tone, such as “clicks”, “pops”, or “thuds”.

It is a further advantage of the invention that a large number of tones and tone durations are produced across and beyond the entire audio range.

It is a further advantage of the invention a sine wave of highly accurate frequency is produced.

It is a further advantage of the invention that a segment of a playing audio stream is replaced with a tone of substantially the same sampling frequency as the audio stream in order to maintain synchronization between audio and video data.

ALTERNATIVE EMBODIMENTS

As previously described, the invention has been described with respect to three embodiments, including a pseudo-code representation (Table 1), a C code implementation (Table 2) and a DSP code implementation (Table 3).

By selection of a different ΔT, a different set of frequencies by the power of 2 may be used to generate a new Table 5 of tone constants. Also, by building a different Table 5 of tone constants, a different scale may be derived, such as one tuned to International Pitch with A

4

equal to 435 cycles per second, or the Scientific or Just scale where C

4

is equal to 256 cycles per second.

It will be appreciated that, although specific embodiments of the invention have been described herein for purposes of illustration, various modifications may be made without departing from the spirit and scope of the invention. In particular, it is within the scope of the invention to provide a memory device, such as a transmission medium, magnetic or optical tape or disc, or the like, for storing signals for controlling the operation of a computer according to the method of the invention and/or to structure its components in accordance with the system of the invention.

Number	Name	Date	Kind
4015220	Kaufman	Mar 1977	A
4056692	Place	Nov 1977	A
4061886	Callahan, Jr. et al.	Dec 1977	A
4150599	Southard	Apr 1979	A
4392406	Fritz	Jul 1983	A
4399535	Southard	Aug 1983	A
4483229	Tsukamoto et al.	Nov 1984	A
4561337	Wachi	Dec 1985	A
4577287	Chrin	Mar 1986	A
4599583	Shimozono et al.	Jul 1986	A
4761751	Canniff	Aug 1988	A
4815352	Tsukamoto et al.	Mar 1989	A
4839842	Pyi et al.	Jun 1989	A
4888719	Yassa	Dec 1989	A
4998281	Sakata	Mar 1991	A
5146418	Lind	Sep 1992	A
5243124	Kondratiuk et al.	Sep 1993	A
5325422	Ladd	Jun 1994	A
5338891	Masubichi et al.	Aug 1994	A
5418734	Kawamura et al.	May 1995	A
5418782	Wasilewski	May 1995	A
5457701	Wasilewski et al.	Oct 1995	A
5524087	Kawamura et al.	Jun 1996	A
5689080	Gulick	Nov 1997	A

System and method for generating and attenuating digital tones

Information

Patent Number

Date Filed

Date Issued

Inventors

Original Assignees

Examiners

Agents

CPC

US Classifications

Field of Search

US

International Classifications

Abstract

Description

Claims

US Referenced Citations (24)