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Dynamic vs. Sampling

Live audio is very dynamic, yet a dynamic range of 130dB is almost impossible for any digital AD and DA converter at the current sampling rates used to replicate. Dynamic resolution is directly related to the bit and sampling rates used in the digital conversion process.

At present, commercial digital audio commonly uses multiple-bit digital standard sampling, (PCM, Pulse Code Modulation). Industry standards have determined that a 24-bit/96 kHz sample rate (which approximately corresponds to 2.8224 MHz 1bit PDM, Pulse Density Modulation, used in SACD), is adequate when professionally converting an audio signal consisting solely of harmonic signal components. In practice however, the audio signal consists of many signals, therefore it is complex and its properties are actually closer to random signals. The spectrum of random signals is infinitely wide, so when converting analog signals to digital, the sampling rate must be as high as possible in order to maintain quality of the transferred signal in full resolution.

At the current sampling rates utilized in commercial audio system design it is evident that their resolution is compromised as the frequency range increases. Much of the detail and ambience captured in the sound is related to this area. A decrease of information due to sampling frequency affects the sound as a loss of brilliance and intelligibility while increasing high frequency sizzle and distortion. In addition, when using DSP, there is additional information loss due to reduced DSP power, precision and limited processing time.

By increasing sampling rates the amount of transferred information increases, but sampling rates cannot be unlimited, so digital audio can never reach the quality of a high definition analog system. It is however necessary in some cases to use digital processing to correct certain aspects of the system in a professional live audio application, KV2 looked at ways of maximising definition and dynamic range in the digital realm.

KV2 undertook a different approach to digital. They looked at an alternate conversion process developed by Sony and Philips called Direct Stream Digital or DSD. The Super Audio CD (SACD) is based on DSD and unlike pulse code modulation (PCM) technology used commonly in pro audio applications and normal CD recordings, DSD technology is based on a one-bit ‘sigma-delta’ converter that produces a stream of pulses. The amplitude of the analog waveform is represented by the density of pulses, and is called Pulse Density Modulation. The resulting digital bit stream is encoded at enormous 2,822,400 samples per second! (2.8224MHz)

Practical listening tests were undertaken by our engineers to determine the minimal sampling frequency required to eliminate any audible information loss. The result saw KV2 design a circuit based around DSD with a sampling frequency of an incredible 20MHz, when using 1 bit sigma-delta PDM modulation. KV2s new digital converter delivers resolution 7 times higher than SACD. A special step compander circuit adds a further 20db of dynamic range to utilize the maximal range of the converter at low levels.




Step compander changes input and output level to increase dynamic range of the converter

KV2 Audio utilizes hybrid signal processing, using the best of analog and digital technology to provide all necessary filtering, equalization and time alignment to their speaker systems. This best of both worlds approach provides unmatched dynamic range and audio reproduction.

Text and photos by courtesy of KV2 Audio. All trademarks, registered trademarks and copyrights are the property of their respective owners.

Impulse Responses

To maintain a high-resolution audio signal, it is also important for the system to maintain the shortest possible impulse response time. Impulse response time is effected by settling time and circuit design in analog and by samplig frequency in digital. In the figure below it is evident that commonly used commercial systems, particularly digital, cannot pass the full resolution of the original signal. Impulse response time is effected in the digital domain by the sampling rate and in the analog signal path by the speed of the electronics (settling time) and control of the acoustic component motion (speaker movement). The change in the original signal caused through poor impulse response creates distortion. 

Systems with long impulse response are unable to transfer high dynamics and high definition signals. SLA systems incorporate hybrid signal processing at an industry leading sample rate of 20MHz and a settling time of 1microsecond (1μs), to ensure audio reproduction with the highest possible resolution and definition.

Impulse time response audio systems to input pulse, level -6dB, duration 3μs

Information transfer of digital and analog sound systems