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  1. #161
    Senior Member Eric Seaberg's Avatar
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    Make sure you're not having any transcoding going on or limiting the bandwidth. Your DTS receiver will want the full bandwidth WITHOUT any interference. This also means you should NOT have the volume control change your digital output as it changes the digital stream.
    Eric Seaberg - San Diego
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  2. #162
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    Quote Originally Posted by Eric Seaberg View Post
    Make sure you're not having any transcoding going on or limiting the bandwidth. Your DTS receiver will want the full bandwidth WITHOUT any interference. This also means you should NOT have the volume control change your digital output as it changes the digital stream.
    Hi Eric, thanks for your reply. The problem is that spdifconvert.py is doing the transcoding. All I want it to do is add the 36 byte (or however many bytes it is) wave header to the raw DTS data file, but instead it's doing frame by frame analysis of the data and playing around with it (e.g. doubling the file size from 15MB to 30MB!).

    I'm playing with the python code now to see if I can add my own "raw" option in addition to the supplied ac3 and dts options such that it will add a wave header to raw files but not modify the data.

    BTW For reference I use the free DTS .WAV samples from Diatonis to check that the receiver is receiving and playing DTS data streams ok.
    Last edited by peterjwhite; 2007-04-22 at 21:07.

  3. #163
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    Quote Originally Posted by peterjwhite View Post
    Hi Eric, thanks for your reply. The problem is that spdifconvert.py is doing the transcoding.
    What Eric means is that after converting it to a WAV, something in the SlimServer -> SqueezeBox2 (or 3) pipeline might be performing additional transcoding that you didn't expect. One source of this is the bandwidth limiting option in SlimServer, which re-encodes to MP3; another is the use of anything other than 100% volume on the digital out.

    Please check the second post of this thread, containing a sub-section titled "Configuring SlimServer".

    All I want it to do is add the 36 byte (or however many bytes it is) wave header to the raw DTS data file, but instead it's doing frame by frame analysis of the data and playing around with it (e.g. doubling the file size from 15MB to 30MB!).
    It MUST convert each frame if the DTS has come from a DVD. Just adding the WAV header is not enough -- the format of the data in the will not be correct if it's exactly the same as the DTS from the DVD. In order to trick the SB2 and the amp into playing this data, we must encapsulate it in an IEC61937 stream, and that means taking each frame and padding it.

    I'm playing with the python code now to see if I can add my own "raw" option in addition to the supplied ac3 and dts options such that it will add a wave header to raw files but not modify the data.
    You are of course free to modify the code as you wish (and if you have success I'd be grateful if you post here again with information) but I honestly believe it won't help. If you've ripped the file from a DVD, and the utility has recognised it as something it can work on, then the correct thing to do is to pad the frames as it's doing.

    BTW For reference I use the free DTS .WAV samples from Diatonis to check that the receiver is receiving and playing DTS data streams ok.
    The files on the Diatonis website use a slightly different form of DTS (a special 14-bit format suitable for packing into WAV in such a way that a DTS-CD can be created, IIRC). They've already been converted to WAV. It's possible that they use less bandwidth than the files you've ripped from DVD, so it's worth checking the SlimServer configuration anyway.

    Quote Originally Posted by peterjwhite View Post
    Hi, I have successfully ripped .dts files from a DVD-Audio but I'm having no success getting spdifconvert.py to create usable .wav files.

    Firstly, I've verified that the .dts files are ok by playing them with dtsdec.exe. Note that the filesize for my test track is about 15MB and the output below from dtsdec shows 15318 frames:
    C:\>dtsdec dvd_ae.dts
    dtsdec-0.7.4-cvs - by Gildas Bazin <gbazin@videolan.org>
    based on the a52dec code from Michel Lespinasse <walken@zoy.org> and Aaron Holtzman
    Stream with high frequencies VQ coding
    15318 frames decoded in 163.23 seconds (93.84 fps)
    When I run it through spdifconvert the result is correctly 15318 frames, but over wireless to my dts decoder it's not recognised as DTS and plays as noise. Note that with the default frame-size=2048 the file size is double of the original, i.e. it has become 30MB. I tried using --frame-size=1006 and while the file was back to 15MB it still played noise. The verbose output is below.
    C:\>spdifconvert.py -v dvd_ae.dts
    Reading input from file dvd_ae.dts
    Writing output to file dvd_ae.dts.wav
    Detected stream type: dts
    DTS frame data: FSIZE 1005
    DTS frame data: NBLKS 15
    DTS frame data: SFREQ 13; sample rate is 48000
    Padding output file to frame size 2048
    DTS length: 15409908; frame length: 1006
    Guessing 15318 DTS frames, WAV length 31371264 bytes
    WAV header does not need fixing (good estimate!)
    Completed transformation. Took 0m02s.
    Any suggestions?
    How long is the track supposed to be? Is it about 2:43? If so, that WAV size seems correct. Is it supposed to be twice that length? If so, it seems wrong! You could try a frame size of 1024 instead of 2048, but I don't know if it will work. If it does, I'd love to hear about it!

    Is it possible to try streaming over a wired connection, in case it's a problem with the wireless? And are you converting to FLAC or streaming the WAV?

    The frame size is chosen by multiplying the number of blocks (NBLKS+1) by 32 (the number of samples in a DTS frame), then multiplying that by the sample size (2 bytes) and the number of output channels (2 -- because we're packing into something that looks like a stereo WAV. The number of channels in the source DTS file is not relevant here AFAIK). The figures you've posted look okay to me. (The fact that FSIZE is smaller indicates, I think, a half-bitrate DTS file.)

    It's generally expected that the output file will be larger than the input file: the WAV will be the right size for 16-bit stereo PCM data, which is what the DTS is pretending to be for streaming to the SB2.

    Please feel free to send me a private message if you'd like me to take a look at the file -- I'll give you an email address that can accept it. I'd be happy to take a look at it when I have a minute (hopefully some time today!).

  4. #164
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    comments on itrax.com and how their files can be played with SS/SB3

    Interested to hear comments on this soon to be released service and how their files can be played with SS/SB3.

    If you read this post from AVS forums it will give you the details of what Dr Mark Waldrep is doing.

    http://www.avsforum.com/avs-vb/showthread.php?t=834714

  5. #165
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    Looks interesting! Thanks for posting the link.

    File formats: 44.1kHz DTS files (for burning to CD) will work with the SB as they are, as WAV or FLAC (or other lossless format).

    48kHz 16-bit DTS and DD files will work if they're suitable for use on a DVD-Video, by converting them (as in this thread). I've never created an audio file for DVD, only ripped one, so that "if" remains to be considered -- but as soon as samples are available we can try it out!

  6. #166
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    Quote Originally Posted by Kiwi View Post
    Interested to hear comments on this soon to be released service and how their files can be played with SS/SB3.

    If you read this post from AVS forums it will give you the details of what Dr Mark Waldrep is doing.

    http://www.avsforum.com/avs-vb/showthread.php?t=834714
    Interesting - I notice he mentions the Transporter pretty early on in the thread:

    Quote Originally Posted by Dr. AIX
    There will also be HD Audio PCM files at 96 kHz/24-bits [both stereo and 5.1 mixes] that can be downloaded and played on through a computer sound card or using dedicated output boxes like the Transporter from Logitech...a high end stereo D/A unit that uses the computer to host the media.

  7. #167
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    Quote Originally Posted by peterjwhite View Post
    Hi, I have successfully ripped .dts files from a DVD-Audio but I'm having no success getting spdifconvert.py to create usable .wav files.

    ...
    An update in case anybody was wondering... we think the problem here is that the playback device is an Airport Express, which (from the little information I can find about this aspect) seems to not support 48kHz files directly. DTS-CD files are 44.1kHz; DTS from a DVD is 48kHz and, if re-sampled, will sound like white noise.

  8. #168
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    I haven't looked at this thread or tried streaming surround files for a while as my wireless network just wasn't up to it.

    But I've just bought a Belkin MIMO router that was able to stream DivX files to my PS3, so I thought I'd give this a go again.

    What I found is that my Sony surround receiver, that's about 8 years old, does not seem to reliably decode AC3 files(converted to WAVs, then FLAC'd), but does handle DTS files.

    I've downloaded AC3 files from Kelly industries to ensure it's not my ripping method, but the amp doesn't like them either. Sometimes the surround light comes on and the display tells me the amp has got a DD5.1 file, but I get no sound. Most times, I just get stereo output.


    However DTS files, either downloaded, ripped from a DTS-CD, or ripped from a DVD work no problem.


    I'm wondering if the SPDifConvert tool that Steve wrote can take an AC3 file but change some header bits so that the amp is told it's a DTS file in the WAV header info. Would that work or are they completely different formats?

  9. #169
    Senior Member Eric Seaberg's Avatar
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    I can easily play DTS or AC3 files through my TP using SPDIFconvert. It works like it's s'pozed to.

    If your surround receiver is REALLY 8-years old, are you sure it's capable of decoding AC3 and not just Dolby Surround? Surround is really just a 4-channel matrix using LCRS phase encoded into a stereo pair. I believe Dolby Digital (AC3) may have been implemented a little more recent than 8-years. DTS, on the other hand, has been around a little longer.
    Last edited by Eric Seaberg; 2008-01-05 at 21:14.
    Eric Seaberg - San Diego
    A.E.S., S.M.P.T.E., S.P.A.R.S.
    eric@seaberg.com

  10. #170
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    No the Sony amp defintely can cope with Dolby Digital. It has no problem whatsoever with my DVD (all 5 or 6 that have been hooked up to it in the past) player. It works fine with my PS3.
    The only thing it doesn't like is Dolby Pro Logic II.

    It will play the spdif'd files but just in stereo. However for some of the test files I downloaded from Kelly Ind etc, it will either not make any sound, but light the surround light and display DD5.1 on it's display or play stereo sound.


    Maybe I'll need to wait until the HDMI amps come down in price before buying a new one. The top of the range Sony amp( http://www.sony.co.uk/view/ShowProdu...inema+Receiver ) has got very good reviews. I'm not looking for audiophile quality, but it does have to be flexible connection-wise.

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