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Dear fellow audio enthusiast, Today I published a first draft of mpd-monitor, a minimal load and realtime monitor of an audiostream as handled by mpd. A screencast says it all: Details and documentation are available at the gitlab repository: https://gitlab.com/ronalde/mpd-monitor/. When you experience bugs or issues, which will be the case, please submit those in the gitlab project. I would like to hear your thoughts. Happy listening and monitoring, Ronald
I have a long history in telecom so I understand the concept of sampling an analog signal and turning it into a digital stream. Perfect example is the early 64K channels in T1's for voice (sample 2x highest frequency and create 8 bit word or each sample (8000Hz * 8 bits = 64K channel). And obviously a digital bit-stream must be sampled at a certain rate to accurately turn it back into analog. But I am confused as to what and how up-sampling works and how it could even create a better quality analog signal from the original digital signal.
Interested in the facts? One of the world’s top converter designers Dan Lavry has written a new paper in simple language to demystify the subject. http://www.lavryengineering.com/pdfs/lavry-white-paper-the_optimal_sample_rate_for_quality_audio.pdf See why many professional engineers still work at 96kHz years after 192kHz became available. Find out why “more” is not always “better!”