Oversampling diagram

Oversampling is a crucial part of a modern DAC, especially the ES9038P which uses 4x oversampling in stereo or 8x in mono mode. The ES9038P has shown its ability to filter out noise that the Play allowed through. We found out this digital noise in the Play, was brought on by having 2 paths to ground. One in the USB cable and one through the chassis ground (supplied by the power supply ground).

Oversampling helps to push noise/distortion outside of the audio band, especially with modern filtering/shaping, in addition to oversampling. With specific shaped filtering you can filter out 50% more or so of the quantization noise than with a plain filter (Source: Art of Digital Audio). 

Oversampling an accuracy

Quantization Noise Diagram

In simple terms, oversampling means to sample at a higher rate than is required by the Nyquist Rate (a little over two times the top input frequency). Quantization (also called signal processing) errors happen when going from analog to digital (ADC). This involves taking a continuous signal and segmenting it into “boxes” or “bins”.

“Such errors are an inevitable result of the classification of continuously varying analog voltages into discrete digital bins. This quantization necessarily adds this noise at the level of ½ LSB (Least Significant Bit).” Pg.694 Art of Electronics Lab Manual 3rd Edition

Luckily for us, the entire process is digital. From hard drive to XMOS to ES9038P (or ES9018K2M for the Play), where finally, digital is then turned to analog (DAC) and sent into an amplifier. Now, if you were using a record player, you wouldn’t need any DAC to play it back, it’s already analog.  

Illustration of Sampling SIgnals
Courtesy of Yamaha

Often after conversion from digital to analog (DAC), flat lines will have a roughness, also called “steppy edges”. This error can be smoothed out using a quality, low-pass filter. If you look on the underside of the Play’s lid, this is what the “LP” means in the diagram. Op-amps are commonly used to create quality low-pass filters, with just a little effort.

However, you can also clean up this noise by simply increasing the sampling rate (usually 24-bit or 32-bit for HiFi DACs). Increasing the sampling rate changes the number of data points per second, thus increasing your sound resolution and accuracy. Here you can see how sampling rate vs. oversampling plays out. Thanks to Rynsin, over on SuperBestAudioFriends Forums. (Click for full thread)

Sampling Rate: Sampling involves breaking the waveform/graph into smaller and smaller chunks. However, the sampling rate must be taken across the horizontal/time axis. (Note: In audio, when signal frequencies begin close to zero, the highest frequency simply is the bandwidth) This tells us the sampling required for a given signal must be dependent on the highest frequency in the analog wave. If the wave is made up of more than one frequency  (such as in audio), it is the highest frequency that we are concerned with (lower frequencies are much easier to sample). So, if you buy a high res PCM/FLAC album that is rated at 192 kHz, the highest frequency you should see on an oscilloscope of the DAC outputs should be right around 192 kHz. In practice, you will not see 192 kHz on the dot.

Curtosey of Yamaha

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