I've been trying to manually do what the PS3 Media Server does: encapsulate DTS audio inside an LPCM so that it can stream to PS3 and be sent to my receiver for the real DTS decoding. It works fine when I use PMS, however, it's slow so I want to do this ahead of time and just stream it. Searching around this forum I found the following directions to use eac3to/spdifer and pcm2tsmu for this task.
In both cases (using spdifer or eac3to) I end up with a file that shows up on the PS3 as Linear PCM but my receiver then outputs static. Any idea what I'm doing wrong?
Thanks for any help!
+ Reply to Thread
Results 1 to 10 of 10
SFAIK, pcm2tsmu is useful only if you prefer the outdated versions of TSmuxer
As for spdifer, the newest version is somewhat buggy, the build below works as it should.
I end up with a file that shows up on the PS3 as Linear PCM but my receiver then outputs static
Last edited by El Heggunte; 18th Aug 2014 at 19:37. Reason: typos, grrrrrrrr
I don't have a PS3 nor will I probably ever, but one thing I do know, when you try to play DTS through something that will not decode it you get static or white noise.
Just putting DTS in a different wrapper or container will not make it anything other than DTS unless you re-encode it.
Yes, but a "dts-wav" will be passed along as pcm by apps/drivers/devices that don't decode dts, and if they're passed to a receiver/amp that CAN, all is well.
Thank you so much. I don't know which of your two suggestions did the trick but it worked. I had tsmuxer version 2.6.11 and updated that to version 2.6.12. Also, I replaced my spdifer with the one in your post. Now it's all working. PS3 is showing a 2ch. LPCM and my receiver is playing DTS and it's all perfectly smooth!
In case it's helpful to others my exact steps where to
1) demux with tsmuxer 2.6.12
2) run spdifer.exe <input.dts> <output.wav> -wav
3) remux with tsmuxer
Thank you all for the quick replies.
UMS but recently it stopped working for me. It only plays a few seconds of audio and no video at all. It used to work just fine before.
I'm actually thinking about just going to mediatomb. Ideally I want a very lightweight DLNA server since my machine isn't really fast enough to do on the fly transcoding/muxing. Any suggestions in that area besides mediatomb?
I just need something to stream the files directly to my client (ps3). If I switch clients in the future and it doesn't like the current codecs I'll just re-rip my DVD/Blu-Ray collection in the new format. It's the price I pay for being cheap and using a nearly 10 year old computer for this
Sounds like you have a crappy computer that has issues,going to another program won't help.Better to buy a better computer.Otherwise you will always have these issues.I think,therefore i am a hamster.
I have made DTS Wav audio CD-r's before but that does not seem to me what he was asking or wanting to do.
And what I said is pretty much still valid.
Like I said, I have never owned a PS3 and I probably never will so I do not know anything about it or software for it.
Yes, it DOES require you to re-encode (unless you use a special "burst-packetizer" insertion app).
No, it IS what he was wanting to do. DTS-WAV (and AC3-WAV) can be done for encapsulating in (and masquerading as) LPCM in a number of applications: CD audio (such as you've done), DVD audio (and DVD-Audio audio), as well as simple PC->SPDIF/TOSLink->Amp playback. For the CD version, you must use 44.1kHz sample rate, for the DVD-Video one, you must use 48kHz, for DVD-Audio or simple PC files, you can use either. IIRC, Minnetonka's SurCode allowed you to do DTS and DTS-WAV, at whichever samplerate was appropriate for the specified destination.
The difference between the compressed audio and the compressed audio-encapsulated-as-wav is that the compressed stream must be less bitrate than an uncompressed PCM stereo stream's bitrate, using the same sample rate as the expected LPCM, and it must be packetized to fall within the standard LPCM frame/sample boundaries. These are done in bursts of data (seen as "noise") interspersed with the remaining silence (to fill the end of the sample boundary) and the receiving/decoding app should be able to buffer the incoming data, ignoring the bits of silence.
Last edited by Cornucopia; 19th Aug 2014 at 22:53.