Need some SPDIF things explained to me.

[Ingram]

Ok currently I have an asus A7N8X Deluxe which has coax out. I am planning to upgrade to a HTIB setup. Most likely this Yamaha one. I have various questions regarding how I will move on from analogue soung to digital.

Currently I am using Zoom Player Pro 4.51, FFDShow to Upscale the image and also decode my audio to stereo. I also use Reclock to occasionaly speed up 23.976 material to 25.000 and sometimes Pal Speed down 25.000 material. Also Reclock is essential for me to keep smooth playback.

Question 1: Dscaler audio has the option for connection type. such as 16 Bit, 24 Bit, 32 Bit, and 32 Bit IEEE Float. All of these work on my setup. Except 32 Bit and 32 Bit IEE Float do not work when Reclock is running. 24 Bit works with Reclock, and the Yamaha website leads me to believe it is 24 bit. So does this pose a problem? Also when I was listening to some AC3 and DTS dvd's Zoom player said the tracks were 16 bit? Which Bit should I be using?

Question 2: Does the receiver have to be getting 48,000hz at all times? Reclock plays around with this figure in realtime making adjustments. When speeding up or slowing down media the difference can be quite large (48,000 to 46,000hz) does this pose any problems?

Question 3: Fusion has the option to pass through to SPDIF. Does this pass through the typicial MPEG audio to my reciever? I played around with it before as best I can without actually being able to hear what is coming through the SPDIF. I played Prime HD with the MPEG track and I was still hearing audio coming through my analogue speakers. But when I switched to the AC3 track sound went dead, I assume because it was going through the SPDIF.

If the MPEG audio doesn't get passed through to my reciever how am I meant to hear the sound? Would I just have to run a stereo coax from my PC to the receiver?

Edited December 23, 2005 by Ingram

[AlexF]

OK, I am not an expert, but here goes.

AC'97 only allows output on SPDIF in two formats - PCM (which is mono/stereo up to 24bit@192Khz, I believe, this is created by the audio render) or datastreams (defined by IEC61937, which includes AC3,MPEG and DTS).

AC3 has three sampling frequencies, 32, 44.1 and 48Khz (non-variable fscod element) and none other. AC3 has variable bit rate (from 32 to 640kb/s) by dictating the number of 16-bit words inside a frame (specified by non-variable frmsizecod), but this at either of the three specified frequencies.

ReClock isn't AC3 decoder/encoder per se, but it can drop/duplicate frames (to keep sync with video), but it must still output at one of the three frequencies, otherwise it won't be a valid AC3 stream.

Edited December 23, 2005 by Alex

[Ingram]

OK, I am not an expert, but here goes.

AC'97 only allows output on SPDIF in two formats - PCM (which is mono/stereo up to 24bit@192Khz, I believe, this is created by the audio render) or datastreams (defined by IEC61937, which includes AC3,MPEG and DTS).

AC3 has three sampling frequencies, 32, 44.1 and 48Khz (non-variable fscod element) and none other. AC3 has variable bit rate (from 32 to 640kb/s) by dictating the number of 16-bit words inside a frame (specified by non-variable frmsizecod), but this at either of the three specified frequencies.

ReClock isn't AC3 decoder/encoder per se, but it can drop/duplicate frames (to keep sync with video), but it must still output at one of the three frequencies, otherwise it won't be a valid AC3 stream.

I've come across a problem with Mp3's not playing ay 44.1kz. They play at 48Kz when I have it set to SPDIF but if it's set to analogue it plays at 44.1kz.

FFDShow is set to Ac3 pass though @ 640Kb so that matches up well. DTS and AC3 is all 48kz so no bit stream problems there. But I do get dropped packets when playing a Mp3 file!

Also, I should be using 24 bit and not 16 bit? When I set FFDshow to Ac3 it doesn't use any of the options for 16 or 24 bit so what would it be using?

[AlexF]

I've come across a problem with Mp3's not playing ay 44.1kz. They play at 48Kz when I have it set to SPDIF but if it's set to analogue it plays at 44.1kz.

FFDShow is set to Ac3 pass though @ 640Kb so that matches up well. DTS and AC3 is all 48kz so no bit stream problems there. But I do get dropped packets when playing a Mp3 file!

Also, I should be using 24 bit and not 16 bit? When I set FFDshow to Ac3 it doesn't use any of the options for 16 or 24 bit so what would it be using?

Windows (Kmixer) is responsible for resampling all non-datastreams to match the resolution and sampling frequency of the SPDIF output (which can be set to a number of values, highest being 24 bits @ 192Khz, or some large value like that) - so, it's not a matter of dropping packets it's a matter of resampling because MP3 is sent as PCM on SPDIF.

AC3 on the other hand is never resampled (because it's a protocol and not samples) and Windows just pass-through it - it's always sent out as it came most common as 16bit @ 48Khz (or other 32, 44.1). (As I wrote above, there's no such thing as 24bit AC3.)

Edited December 23, 2005 by Alex

[Ingram]

Are you sure about dropped packets there for mp3? I was playing an mp3 in Zoom player as a test. Using FFDShow to decode the audio and pass it through to Ac3 (SPDIF) I also had Reclock running which was reporting dropped packets.

Either it is the case, or it was reclock doing the dropping and I infact won't get dropped packets when I just use Winamp.

Speaking of which will Winamp automaticially pass it's audio through to SPDIF?

Also does it make a difference if my Pc us passing it through using Waveout or Directsound? I think I can do either but what's the difference.

Thanks for all your help. I have been asking questions on bigger forums and no one seems to want to help a noob

Edited December 23, 2005 by Ingram

[AlexF]

Are you sure about dropped packets there for mp3? I was playing an mp3 in Zoom player as a test. Using FFDShow to decode the audio and pass it through to Ac3 (SPDIF) I also had Reclock running which was reporting dropped packets.

Either it is the case, or it was reclock doing the dropping and I infact won't get dropped packets when I just use Winamp.

This doesn't really make sense. ReClock attempts to match video and audio... MP3 is audio-only, so, why would it drop/repeat anything? Besides, MP3 would be decoded by an application to PCM-like samples prior to rendering - there are no frames!

Speaking of which will Winamp automaticially pass it's audio through to SPDIF?

Depends on what you told the audio renderer to do.

Also does it make a difference if my Pc us passing it through using Waveout or Directsound? I think I can do either but what's the difference.

Waveout is older and seems to be more stable.

Thanks for all your help. I have been asking questions on bigger forums and no one seems to want to help a noob

Like I said, I'm no expert, just a little here and there.

Edited December 23, 2005 by Alex

[digitaladvisor]

Hi Ingram

Before those others dig themselves deeper into the SP/DIF standard there are INDUSTRY PROTECTION standard compliance rules applied to PCs and DVD players -

When any COMPLIANT Dolby Digital type is detected - ranging from pro-logic through to 5.1 the Sound being delivered via analog is disallowed - it goes into MUTE state but SP/DIF remains live in bit stream.

When A broadcaster is NOT doing ANY dolby digital standard THE SP/DIF works along with analog outputs.

Turn SP/DIF exclusive off - This tells the softare players to down convert DD 5.1 to stereo - on some sound cards the SP/DIF will then work in a down converted PCM stereo version.

EXCLUSIVE SP/DIF setup via drivers/softwares implies an "anticipated" DD sound stream and will always MUTE analog outputs.

DA