What do you think when you read statements like, “I’ve felt from the start in the 80’s that there was something inherently wrong with PCM audio? For some reason (and I’m not alone) I cannot stand listening to PCM recordings for more than 5-15 minutes without experiencing some sort of inner distress”. Do you believe this? I’m not sure why this nonsense persists. There it was in a review of a program called “Master Class”. Here’s how the software is described on the Daniel Hertz SA (Mark Levinson’s company site):
“Daniel Hertz Master Class is a music player software for Mac with revolutionary capabilities. For the first time in audio history, listeners can have the sound and feeling of SACD and the best analog master tapes from PCM digital sources including CD and compressed music downloads.”
I’m already on high alert after reading this. As the producer of PCM recordings with more dynamic range and higher fidelity than SACD (DSD 64) and analog tape, why would I possibly want to diminish the quality of my recordings by running them through Master Class? It’s crazy to think that this process or any other process that changes the sound of my masters would be an improvement. Since when did SACD and analog tape become the ultimate fidelity standards?
And here’s an overview of its features:
• The infamous A+ algorithm that smoothens the PCM step function waveform.
[NOTE: PCM is not a step function waveform…the output of your DAC is a smooth, accurate, reflection of the analog waveform that entered the ADC at the source end of the process! Maybe the algorithm is “infamous” because it’s snake oil.)
• A six band graphical EQ that let’s you refine the tonal balance (frequency response) of the recording and match it to your playback system.
And finally, the user manual explains the “concept” behind the “infamous” A+ algorithm this way:
“The analog waveform is smooth. The PCM digital waveform is a step function that only approximates a sine wave. The human brain, like the sine wave, is from nature. The PCM digital audio step function waveform is something made by technology that does not exist in nature. A+ is proprietary algorithm that fills in the ‘steps’ to smoothen the waveform so the brain responds more like if it were a sine wave. A+ uses the original signal to generate the ‘fill’ so there is no non-musical information being added. A+ adds no noise or filtering.”
Mark Levinson is a well know fan of DSD and obviously doesn’t understand how PCM digital encoding works if he believes that a “PCM digital waveform is a step function that only approximates a sine wave”. First, there is no such thing as a PCM digital waveform. The output of an analog to digital converter is s stream of ones and zeros that are stored on some sort of media. It is not a waveform…the stream is a collection of data. If the source is a sine wave, then the output from the DAC at the other end of the process is an accurate recreation of that sine wave. There’s no approximation…it’s like using a set of blueprints to construct a building. Is the resultant structure and approximation of the original design?
The type of thinking described by Mark Levinson’s website incorrectly employs analog signal thinking to digital methods…they are two discretely different ways of dealing with audio. It sounds like his “unique” process is upsampling or using interpolation to “fill” in the data stream with more samples. This technique provides the designers of the converters to use better filters but it doesn’t fundamentally change the fidelity of the recording.
I’m headed down to the Newport Beach Audio Show shortly…hopefully, I’ll see some of you there. And at 2 pm, I’ll be one of the guests on Scott Wilkinson’s Home Theater Geek show once again.