Maintaining Audio Quality in the Broadcast Facility 2011 - Orban

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in the Broadcast/Netcast Facility 25 There are a number of sound card and USB audio devices that suffer from bit-slip due to a reversed left and right audio clock. The result is digital audio that is not correctly time-aligned, which causes an interchannel phase shift that increases with frequency. Consequently, the sum of the left and right channels does not exhibit a flat frequency response. The amount of attenuation at a given frequency will depend upon the sample rate. A one-sample slip at 32kHz sample rate will produce a notch at 16kHz and almost 6dB of loss at 10kHz; 44.1kHz will produce almost –3dB at 10kHz and–6dB at 15kHz; 48kHz will produce –2dB at 10kHz and –5dB at 15kHz. A one-sample slip is definitely audible in the mono sum, so devices with this problem are inappropriate for broadcast audio applications, especially for mastering a library. Many of these devices were based upon a Texas Instruments USB Codec chip that had its hardware clock reversed. TI has acknowledged the problem and has released revised parts. However, there are still many audio interfaces and codecs in use with this problem. This hardware should be scrupulously avoided. Level metering in sound cards is highly variable. Average, quasi-peak, and peak responses are all common and often inadequately or incorrectly documented (see Metering on page 26). This is relevant to the question of line-up level. EBU R68 specifies reference level as –18dBfs, while SMPTE RP 155 specifies it as –20dBfs. Unless the sound card’s metering is accurate, it is impossible to ensure compliance with the standards maintained within your facility. Many professional sound cards have adequate metering, while this is far less common on consumer sound cards. Further, consumer sound cards often cannot accommodate professional analog levels, balanced lines, or AES/EBU inputs and outputs. Bit Accuracy Digital systems that do not have sample rate converters in their signal path are considered synchronous if there is no further digital signal processing. A master clock signal, typically in wordclock or AES11 format, is required for all devices in the audio signal path to remain in sync. To be bit-accurate, a system must be synchronous and must not change the word length of the audio data being conveyed. Digital systems that have sample rate converters in their signal path are not bitaccurate and are considered asynchronous even though they can also be locked to a master clock. The advantage to asynchronous digital audio is that just about anything can be connected to just about anything else without difficulty. However, there are cautions. Computer audio can be very confusing, especially when one is trying to achieve bitaccurate audio. Because audio has many different sources and destinations, computer operating systems contain software sample rate converters for both record and play to make things easier for the average user. For the professional, this can cause performance problems and sample rate conversion can occur when it is not wanted or expected. To work around this problem, most professional users use the ASIO interface and drivers. However, not all audio applications support ASIO. Furthermore, if recording or encoding from another application is required, this is not supported by ASIO.
26 Maintaining Audio Quality Fortunately, Microsoft Windows XP has a high quality sample rate converter in the audio stack for both record and play. To achieve all of its potential quality, adjust XP’s Sample rate conversion quality to Best. (The control is located in Sound and Audio Device Properties>Audio>Sound Playback>Advanced>Performance.) To prevent sample rate conversion from occurring, be certain that source sample rates match record sample rates and that play sample rates match destination sample rates. If sample rate conversion is unavoidable, Windows XP will provide a highquality conversion. Microsoft implemented a new audio stack in Windows Vista/7. Legacy audio applications use the emulation mode of this new audio stack. Unfortunately, Microsoft seems to have broken the record sample rate converter, which is to be avoided at all costs if quality is of concern. The play sample rate conversion appears fine. We have contacted Microsoft regarding this issue. As of this writing, we have received absolutely no comment. Metering One of the most misunderstood details of audio is exactly how to measure levels, and how analog levels relate to digital levels. The diagram below shows the calibration and level relationships between the following meters, where all meters are displaying a sinewave at SMPTE reference level: � True Peak-Reading Digital Level Meter � VU Meter (ANSI) � Peak Program Meter (EBU IEC 268-10 IIB � Peak Program Meter (UK IEC 268-10 IIA) � Peak Program Meter (Nordic IEC 268-10 I) � Peak Program Meter (DIN IEC 268-10 DIN 45406)
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Maintaining Audio Quality in the Broadcast Facility 2011 - Orban

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