TDA1307 データシートの表示（PDF） - Philips Electronics

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TDA1307

High-performance bitstream digital filter

Philips Electronics 

Philips Semiconductors

High-performance bitstream digital filter

Preliminary specification

TDA1307

Digital silence detection

The TDA1307 is designed to detect digital silence

conditions in channels left and right, separately, and report

this via two separate output pins, one for each channel,

DSL (pin 11) and DSR (pin 12). This function is

implemented to allow for external manipulation of the

audio signal upon absence of program material, such as

muting or recorder control. The TDA1307 itself does not

influence the audio signal as a result of digital silence; the

sole function of this block is detection, and any further

treatment must be accomplished externally.

An active LOW output is produced at these pins if the

corresponding channel carries either all zeroes for at least

8820 consecutive audio samples

(200 ms for fs = 44.1 kHz).

The digital silence detection block receives its left and right

audio data from the error concealment block (implying that

a digital mute action will produce detection of a digital

silence condition), and passes it unaffected to the next

signal processing stage, the de-emphasis block.

De-emphasis filter

The TDA1307 incorporates selectable digital de-emphasis

filters, dimensioned to produce, with extreme accuracy,

the de-emphasis frequency characteristics for each of the

three possible sample rates 32, 44.1 and 48 kHz. As a

20-bit dynamic range is maintained throughout the filter,

considerable margin is kept with respect to the normal CD

resolution of 16-bit i.e. the digital de-emphasis of TDA1307

is a truly valid alternative to analog de-emphasis in

high-performance digital audio systems.

Selection of the de-emphasis filters is performed via the

microprocessor interface, bits DEMC1 and DEMC0, for

which the programming is given in Table 3.

Oversampling digital filter

The oversampling digital filter in the digital audio

reconstruction system is of paramount influence to the

fidelity of signal reproduction. Not only must the filter

deliver a desired stop-band suppression while sustaining a

certain tolerated pass-band ripple, but it must also be

capable of faithfully reproducing signals of high energy

content, such as signals of high level and frequency,

square wave-type signals and impulse-like signals (all of

these examples have their counterparts in actual music

program material). Filters optimized only towards

pass-band ripple and stop-band suppression are capable

of entering states of overload because of the clustered

energy content of these signals, thus introducing audible

degradations in processing the mentioned types of

excitations. To dimension a high-fidelity digital filter, a

balance must be established between filter steepness and

overload susceptibility.

The oversampling digital filter function in the TDA1307 is

designed, in combination with the noise shaper, to deliver

the highest fidelity in signal reproduction possible. Not only

are stop-band suppression and pass-band ripple

parameters to the design, but also the prevention of

detrimental artifacts of too extreme filtering: impulse and

high-level overload responses. The outcome is a patented

design excelling in natural response to most conceivable

audio stimuli. It is realized as a series of three half-band

filters, each oversampling by a ratio of two, thus achieving

an eight times oversampled and interpolated data output

to be input to the noise shaper. Each stage has a finite

impulse response with symmetrical coefficients, which

makes for a linear phase response. Filter stages 1, 2 and

3 incorporate 119, 19 and 11 delay taps respectively.

To maintain an output accuracy of 20 bits, an internal data

path word length of 39 bits is used to supply the required

headroom in multiplications. Requantization back to 20-bit

word length is performed by noise shaping (thus effectively

preventing rounding errors in so far as they have effect in

the audio frequency band), at the output of each filter

section.

The successive half-band filter stages are, for efficiency,

distributed over the audio data processing path:

DC-filtering, peak value reading and volume control are

performed between stages 1 and 2 (the 2fs domain).

DC-cancelling filter

A mechanism for optionally eliminating potential DC

content of incoming audio data is implemented in the

TDA1307 for three main reasons. Most importantly

because it is called for by the implementation of volume

control in the TDA1307. An audio signal that is to be

subjected to volume control (multiplication by a controlled

attenuation factor) should be free of offset, otherwise the

controlled multiplication will produce the undesired side

effect of modulating the average DC content. The second

reason is supplied by the implementation of audio peak

data read-out in the TDA1307. As the peak value is

obtained from the absolute value of the audio data

referenced to zero DC level, its accuracy is impaired by the

presence of residual DC information, progressively so for

lower audio levels. The third reason is brought about by

application of the noise shaper. To optimize the dynamic

behaviour of the noise shaper especially for low-level

signals, it is supplied a predefined offset, sometimes

referred to as DC dither. Taking no precautions against DC

1996 Jan 08

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