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Clean up the Audio
Another important factor regarding the coded system is headroom. Digital systems have an absolute maximum ceiling of 0dBfs. Theoretically, audio levels for transmission should be able to be set right up to this level. But, depending upon the encode/decode implementation, overshoots may occur. This is not consistent from codec to codec, but more so due to the implementation of the codec by various manufacturers. Additional input low-pass filters in the encoder may cause headroom difficulties. A well-designed encoder will ensure that any added input filter possess the same headroom as the system without generating overshoot that reduces headroom. Note: Most filter overshoot ranges from 2dB - 3dB, but can exceed this amount depending on filter characteristics.
It would be wise to test any codecs within a specified infrastructure to make sure that 0dBfs, is attainable without system overload or clipping. For this reason, setting the absolute peak level 2dB - 3dB below 0dBfs offers insurance to avoid clipping.
HD Radio has the capability to broadcast multiple content streams within the 96kb/s digital channel. Multicast requires the use of lower bit-rate audio coding. It is possible that extremely low bit-rate audio channels will exist, and require dynamics processing capable of consistent sound quality that yields low, or no sonic artifacts.
For those who wish to tweak on their own with existing processing equipment, the following should be observed:
Avoid dense processing that contains fast limiting time constants. Try to reduce the attack time on functions when 5dB, or more, depth-of-compression is desired. This will reduce upper frequency processor induced IMD.
Make sure the coding system provides full headroom. If the system clips on its own before 0dBfs, then reset the maximum input level to avoid system headroom problems.
Low bit-rates will benefit from bandwidth control. A static low pass filter will reduce artifacts. The tradeoff will be perceived high frequencies vs. quality. A specialized processor for coded audio will offer some dynamic method to accomplish this.
Do not use any final limiter that contains a clipper. The THD generated by the clipping function will cause more trouble than it's worth. Precision peak control is needed in the coded system. As mentioned prior, specialized processing for this medium will provide a look-ahead limiter to accomplish this task. If these four steps are followed, improved coded audio will result.
Codecs and clipping
Sound media require peak control to avoid loss of headroom and eventual system distortion. Precision peak limiting accomplishes this. Hard limiting or peak clipping is used in conventional broadcasting, and it works quite well. The method does not technically degrade the system. (Overuse of final limiting is a subjective adjustment, and too much can degrade performance.) Suffice it to say that hard limiting does work as a precision peak controller within FM stereo and AM transmission.
The coded path offers a different set of challenges. It is not possible to overmodulate the system, as there is a precise peak ceiling of 0dBfs. Precision peak control is required, but the conventional method of clipping creates systemic problems, and those occur as aliasing products within the encoder. Figure 4 is an example of what happens to a 2kHz tone when clipped and 15kHz low pass filtered in a conventional audio processor used for FM stereo and passed through the HD Radio codec. This problem is consistent with other codecs too.
The cluster of energy that appears around 15kHz is aliasing components. These were caused by the 2kHz clipped signal from a conventional audio processor as the hard limited signal was routed to the codec. This is proof that all peak limiting for coded audio must use a limiting means that is void of THD content. Clipped waveforms are exceedingly high in THD. This is why the use of look-ahead limiting is the preferred mechanism for encoders. This style of limiter yields very low THD, and will not alias the system.
For reference purposes, Figure 5 is the same signal, prior to the codec. Notice how the odd harmonics line up as would be expected from a clipped waveform. The added strange content that appears around 15kHz in Figure 4 is what exaggerates coding artifacts when conventional style processing is applied to coded audio.
Research, testing, development and hopefully sound reasoning offered here now explain why coded audio performs as it does. Various signal processing and conditioning means can be used to bring life to coded sound. The test results illustrated here reveal that conventional compressors and limiters exaggerate artifacts. While signal processing, conditioning and peak limiting is required for coded audio, the processing must employ methods that do not contribute additional distortion aspects, as this is what degrades clarity and quality at low bit-rates, and sometimes even at moderate to higher rates.
Foti is president of Omnia Audio, Cleveland.
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