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View Full Version : "no valid signal" with ac3 spdif passthrough on cmi8738


loki03
19th January 2006, 15:23
hy!
i got a dvd-player-tuner-speaker combination device, a philips lx7000sa. it plays dvd´s with ac3 sound perfectly.

i connected my pc soundcard to it via spdif out of my soundcard, a philips psc604, its a cmedia 8738 chip driven thing.

i can hear everything over this connection: dolby prologic works like a charm, pcm.

the manual of the dvd player says: you cannot input sacd or mp3 via digital-in. it wont play it. use a disc instead. it doesnt say: no ac3 in on the digital-in.

but the device goes to "no valid signal" when i switch to spdif pass through of ac3 sound. when i switch back to dolby prologic or pcm or anything else it immediately goes on playing that back.

i am sure i have no software problem, since its the same effect with the nvidia forceware audio driver (which has even 4 modes for spdif (allow 24 bit, spdif 1, spdif 2), the cyberlink power-dvd spdif feature, the ac3filter spdif passthrough or vlc´s spdif feature.


i connected the digital-in to the soundcard via a 5 m long normal cinch audio cable.


does anyone know, if my dvd-player is programmed not to decode external ac3, or if the cable is the cause ? i dont think its the cable, since raw pcm with 24 bits works perfectly and i think this needs a much higher bandwith on the cable than an 384kbps-ac3 stream. i cant believe its the tuner-device, why should it block ac3 streams, the manual doesnt state anything about this. what it does say is that the device will respect copyright flags (dont know if this only concerns cd).


the soundcard control app just has an option "spdif out on", nothing else. whilst spdif out is working it is also playing back sound on the ananlog out. whilst spdif passthrough is activated in a software player no analog out is played.

what can i do?
thx!

Rockaria
20th January 2006, 00:51
The Philips LX7000SA Home Audio System (http://www.amazon.co.uk/exec/obidos/ASIN/B00006RCSN/026-8523061-7289248) has Dolby Digital 5.1(aka ac3) decoder in it as well as DTS, Dolby Prologic and MPEG decoding by the spec. So it should have no problem decoding the ac3 in-stream.
Compared to the receiver, the Philips Dynamic Edge 4.1 PSC604 Sound Card (http://shopping.yahoo.com/p:Philips%20Dynamic%20Edge%204.1%20PSC604%20Sound%20Card:1991006258:page=details;_ylt=AhjWOiM7LKiVqgYK6ruWF7OrmbsF;_ylu=X3oDMTBiZ2o2Y3ZwBHNlYwNzaWJzcGVj) looks very poor, but the digital quality will be affected little except the sync-failure because of the unsupported stream properties(sample rate & size : usually 48k 16bit is preferred for cheaper cards & onboard codecs) by the audio driver.
The C-Media 8738 Audio S/PDIF In/Out Technical Data (http://www.cmedia.com.tw/doc/CMI8738_spdif.pdf) says it supports both 44.1k and 48k sampling rate. Even if it is true, based on my experience, 44.1k isn't supported by many audio drivers.

Check & set the sampling rate to 48k before feeding to the spdif-out and see if it solves the problem. The ffdshow dsfilter has complete DSPs for resmpling-ac3_encoding-spdif_passthrough.

loki03
20th January 2006, 02:45
wow, great tip! thank you! i think this brings me much closer to a solution, although things got real tricky now:

zoomplayer, reclock, ffdshow audio
thx optimode sourround test.vob
encode_ac3_samples.ac3 (both 48khz)

when i choose as codec "spdif" under Codecs in ffdshow audio the receiver states "no valid signal", no matter if audio is resampled or not. reclock states: x drop y repeat (so its passthrough, otherwise reclock would use frequency correction)

when i choose liba52 as codec it plays as usual- no passthrough. i can resample audio to whatever i want with ffdshow. it plays. but the receiver decodes dolby prologic of course.

so i choose liba52 and check "ac3" under the input/output tab of ffdshow. no more audio (so its passthrough now?!). i check resample. still no audio. i change the samplerate to 44100. no audio. 22000. gosh! audio! any sample rate plays, but 44.100 and 48.000. 44.101 and 48.001 play!
90000 plays choppy...42356 plays perfectly. BUT reclock doesnt show "x drop y repeat", it shows audio correction x hz and sample rate near the one i resampled to. and :scared: its dolby prologic whats coming out of the speakers- no discrete rear left or right, just monoaural rear ....


i dont really get it - does checking ac3 unter the input output tab mean ac3 passthrough? if yes, how can reclock adjust the frequency, as it cant decode ac3 and has to drop repeat frames with this audio format? or is spdif under the codec tab needed and it does do resample, but my receiver/soundcard still dont get it? if it is so, why dont i have audio when checking ac3 in the i/o tab and dont resample? or does checking ac3 in i/o mean ffdshow creates ac3 of the stream, gives it to directshow, which decodes it and passes it to reclock?

thank you so much for your help!!!

Rockaria
20th January 2006, 04:43
The 'spdif' in the codec setting is for pure spdif-passthrough. It even bypasses any DSPs in the filter-chain of the player in addition to the APU(system mixer) DSPs & DAC.

The ac3 output mode does the ac3 dynamic encoding(upto 640kbps) and passthrough together. So it is affected by any DSPs enabled including the resampling.

Your case seems to be the clock generator problem for the CMI 8738 codec which is containing the spdif transmitter.

If the card does not solve the problem, try with a different card or onboard codec whichever has the spdif-out socket.

Good luck.

[edit]
What I am noticing in the CMI Mixer advanced controls dialog box is the SPDIF-IN Loopback to SPDIF-OUT which should be off for normal spdif in-out operations.

loki03
20th January 2006, 11:54
hy, thanks, i tried again with a daily build of ffdshow, it nows shows what it is outputing in the info section, so i dont have to rely on reclock for that info. and what i found is that ffshow only outputs ac3 if the sample rate is unchanged or if it is resampled to 44.1khz. with any other sample rate (like 44.101 or 22.000) it switches to pcm output (reclock info leads to the same conclusion, with sample rates 44.1 or 48. it shows ac3 passthrough, with any other it shows pcm and does rate adaption.).
so i think ffdshow cant create any ac3 other than with sample rates 44.1 or 48khz. but in both cases the receiver dislikes it.

for the loopback i checked every combination it doesnt change anything. i use the psc604 driver and interface, since the generic cmi 8738 driver greyed out the spdif enable option.

so, either my soundcard driver is not capable of handling ac3, or the soundcard itself awaits pcm all the time, or my receiver cant handle external ac3...

which low-budget soundcard would you suggest? thanks!

tebasuna51
20th January 2006, 13:24
what i found is that ffshow only outputs ac3 if the sample rate is unchanged or if it is resampled to 44.1khz. with any other sample rate (like 44.101 or 22.000) it switches to pcm output (reclock info leads to the same conclusion, with sample rates 44.1 or 48. it shows ac3 passthrough, with any other it shows pcm and does rate adaption.).
so i think ffdshow cant create any ac3 other than with sample rates 44.1 or 48khz. but in both cases the receiver dislikes it.
ffdshow or any other encoder. In ac3 only 32, 44.1 or 48 KHz are allowed.

loki03
20th January 2006, 13:49
ok, i mailed to philips asking if my receiver can decode external ac3. meanwhile i found this thread, where one person states that asus onboard cmi8738 can only output stereo, perhaps my card cant understand ac3
http://archive.avsforum.com/avs-vb/showthread.php?s=&postid=2012516#post2012516

Rockaria
20th January 2006, 14:04
for the loopback i checked every combination it doesnt change anything. i use the psc604 driver and interface, since the generic cmi 8738 driver greyed out the spdif enable option.
There seems to lie the reason, the driver issue. Either find a correct driver or buy a new card or a mobo.

If you just want the 7.1(5.1) spdif out CHAINTECH AV-710 8 (7.1) Channels PCI Interface Sound Card - Retail (http://www.newegg.com/Product/Product.asp?Item=N82E16829120103) or SABRENT SND-P8CH 8 (7.1) Channels PCI Interface Sound Card - Retail (http://www.newegg.com/Product/Product.asp?Item=N82E16829130002) looks like a cheapest option. But if you want an all-round multi channel spdif-out, HDA Mystique 7.1 Sound Card (http://compreviews.about.com/od/multimedia/gr/HDAMystique.htm) or similar will be the answer.

http://www.turtlebeach.com/site/products/soundcards/mtgoddl/indetail.asp
http://sounden.terratec.net/modules.php?op=modload&name=News&file=article&sid=237
http://www.cmedia.com.tw/product/CMI8788.htm
http://www.m-audio.net/products/en_us/Revolution71-main.html

On-board APU also has relatively cheaper solutions like azalia for intel boards, which is recently adopted by ATI for it's new SB chipset(SB 450) for AMD platform (http://www.vr-zone.com/print.php?i=2904).

Personally, I prefer my AN7 which has the soundstorm(5.1 DD DICE) with built-in spdif i-o socket & ALC658 codec.

loki03
20th January 2006, 14:14
thanks a lot! i´ll wait some days for the philips reply then i go for a new soundcard. one last thing: is any of these soundcards able to record spdif in (my tuner has spdif out and that would allow me to make hq radio recordings) thx

Rockaria
20th January 2006, 14:43
For the spdif-in PCM stereo, almost any card including the on-board codec with spdif-in socket can lively recode/capture even with the sound recder.

For ac3 streams, many cards including the onboard codec like my AN7 & ALC658 can capture the ddwav(ac3 5.1ch 48k wrapped in 44.1k pcm) with sound recoder or capture device used in graphedit, which can later be played by VLC or stripped to AC3 format with besplit.

I've heard some live dd recoding with audigy & kx driver solution which you can find more information here (http://forum.doom9.org/showthread.php?p=636376#post636376) and here. (http://forum.doom9.org/showthread.php?s=&postid=338286#post338286)