<div class="gmail_quote">On Tue, Feb 9, 2010 at 6:51 PM, feadog <span dir="ltr"><<a href="mailto:feadog@orcon.net.nz">feadog@orcon.net.nz</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div class="im">sorry, spdif is next years project, but I wonder why ac3 should be<br>treated differently than mp2.<br></div></blockquote>
<div>AC3 is not just a bunch of raw samples - it's an encoded format. SPDIF only supports 2 channel 16 bit PCM samples at specific rates (48, 44.1, 32kHz). Both AC3 and DTS are wedged into that format so you can't look at the 2 channel SPDIF data and draw any conclusions about the multi-channel content it represents. Try outputing DTS by SPDIF to a receiver that doesn't understand DTS and you'll see what I mean! The only way to manipulate the volume (or anything else) is to decode the signal to descrete channels, do whatever you want to those channels, and then re-encode it afterwards. I think that can be done, but it's not the same as manipulating PCM data like the setup you've got does.</div>
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<div>Cheers,</div>
<div>Steve</div></div>