Date: Sun, 16 Sep 2012 12:01:09 -0400 (EDT) From: Stuart Barkley <stuartb@4gh.net> To: freebsd-multimedia@freebsd.org Subject: Re: Sound system developement question Message-ID: <alpine.BSF.2.00.1209161053130.46761@freeman.4gh.net> In-Reply-To: <50547CA3.2010209@gmail.com> References: <4B739CF4-5D1D-4FE0-83FD-6987DCB40866@gmail.com> <CABzXLYNvsJm6QKPJga2FYFm_ssDW2cz9ZVrDfx5-DeMFW=0LAg@mail.gmail.com> <201209151453.41118.hselasky@c2i.net> <50547CA3.2010209@gmail.com>
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This message is in MIME format. The first part should be readable text, while the remaining parts are likely unreadable without MIME-aware tools. --4280880523-1718368019-1347809008=:46761 Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-15 Content-Transfer-Encoding: 8BIT Content-ID: <alpine.BSF.2.00.1209161124391.46761@freeman.4gh.net> On Sat, 15 Sep 2012 at 09:03 -0000, Roberth Sjonøy wrote: > Bitperfect function does that and much more, just what I want but > bitperfect is one thing bits are perfect even if there is a lot of > jitter. Do you think you are hearing a problem? Is your D/A indicating problems with jitter? Your initial question is good. It would be a good regression test to send data between two systems via a digital audio connection and ensure that the transfer is bit-perfect. This would prove that the system can operate without bit loss. However, this does not address hardware quality issues in other systems. I think you are either over thinking things or have been reading some pretty old information. Jitter should not be a problem for most modern digital audio connections. This is more information to not take religiously. The details can be complex and some is speculation on my part. Older systems (> 10-15 years) could have problems and occasional buffer underruns where more common (careful system tuning could address most issues). On a properly functioning modern system FreeBSD will keep the byte stream full to the audio chip. Buffer underruns might still occur with modern systems with incorrect tuning but should not happen with default system parameters. Once at the audio chip, FreeBSD has little control over things and you are at the mercy of your specific hardware. The jitter is in the analog wave form of the digital signal. You can be subject to jitter on the clocking source (quality of sound 'card'), jitter on the transmission cable (quality of cable and connectors), jitter in the D/A (quality of hardware), etc. It sounds like you have an older D/A converter (not an oversampling converter) where jitter on the S/PDIF cable matters. Be sure to use quality cables with solid connectors. If you are very concerned about this, buy "monster" brand cables. Consider replacing the D/A if jitter is still causing you problems. You only talked about jitter. If you are also seeking low latency (delay between analog in to analog out), that is a different story and can still cause issues as you reduce the audio buffering throughout the system. Stuart Barkley -- I've never been lost; I was once bewildered for three days, but never lost! -- Daniel Boone --4280880523-1718368019-1347809008=:46761--
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