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17th April 2006, 00:58 | #22 | Link |
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Ok, I'll try to do so in the coming days. In the meantime, reread the guide. If you follow the steps exactly as written, you should be able to succeed. The Surround Encoder is only available when you are in Multitrack mode, not in Edit mode. If you were able to follow the guide to the point where you have the effects loaded in multitrack view, then all you have to do is press Ctrl+E or select "Surround Encoder" from the "View" menu. If it's not in the View menu, then you are not in the right mode. In such a case, choose "Multitrack" from the View menu.
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25th April 2006, 06:37 | #24 | Link |
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@ ursamtl
i'm getting wonderful results with your guide! two question,i can't find the answer anywhere: 1- i extracted some ac3 files from dvds,they are 48000-448k(5.1). how and where can i know how many bits(16,24 or 32) this tracks have? 2- if i encode the ac3 at 448k,means that each channel have 74,666k? (448/6=74,666) thank you so much! |
25th April 2006, 12:53 | #25 | Link |
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Thanks for the positive feedback on the guide. I'm glad to hear it's working for you. Most AC3s I've encountered have 16-bit resolution. I believer 24-bit AC3s are possible, but not 32. As for finding out, I'm not sure, perhaps Foobar 2000 will tell you. I know it does for wave files. I'm not sure whether the AC3 spec divides its bandwidth equally among the 6 channels. You'd have to check through the technical literature over at www.dolby.com or perhaps someone here will know. I'm by no means an expert.
Regards, Steve. |
14th May 2006, 00:01 | #26 | Link |
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hy Steve,..this is the best answer that i could find:
'The central philosophy behind AC-3 is that all channels should be compressed together as an ensemble, where the total bits that can be accommodated by the media (which in this case is film) is distributed among the channels. The input to the AC-3 encoder is six-channel PCM audio (16 to 24-bit resolution and 48kHz sampling rate). The first step is to transform each of the channels from the time to the frequency domain, using Time Domain Aliasing Cancellation (TDAC). Blocks of 512 samples, or 10.7ms of audio, are normally used to yield 256 spectral coefficients. However, when a transient signal is detected, the block size is reduced to 5.4ms duration to minimize pre-echo.' 'A block diagram summarizing the processes for AC-3 is shown in Figure 4. Input is six-channel PCM (solid black arrows), which is then transformed into the frequency domain (F), with block size determined by the transient detector (T). The binary number spectral coefficients are converted to floating-point and coded (C) by expressing exponents (gray dotted arrows) with the lowest bit resolution necessary and re-quantizing mantissas (Q; black dotted arrows) via bit allocation (A). The output compressed data stream (black dashed arrow) consists of the mantissa and exponent information from all of the channels, plus auxiliary data for exponent coding, coupling coefficients (gray dashed arrow), bit allocation, etc. It should be noted that several other coding strategies than those just described are implemented to achieve low data rates; AC-3 is by far the most complex of the three codecs used in digital film sound. The steps for decoding the data are essentially the reverse of those for encoding, requiring the auxiliary data for parameters and information on reconstructing the original channels.' http://www.surroundmusic.net/articles/packing2.html maybe KpeX can write better expanations with more details about the distribuition of bitrates in ac3 inside audio faqs(i will). thanks! |
3rd October 2006, 01:00 | #27 | Link | |
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Quote:
hi Steve. i did this some time ago but trust me, i forgot how i did and lose my backup. now with a new decoder i want to test in cdrw but i don't know and can't find about AC3wave or DTSwave. can you tell me please what program i use to encode as AC3Wave or where i can read about it? thanks in advance! Last edited by raquete; 3rd October 2006 at 02:48. Reason: found solution! |
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26th November 2006, 01:22 | #28 | Link |
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I made a DTS CD and it sounds great! No complaint at all. But now I tried to work with a video file (MPEG 2) and I edited the audio according to your guide (the sound is fine, DTS), but when the program renders audio+video to prepare for the DVD files, the sound is always played as a long hiss. If I put just audio into the program (Architect), the DVD plays fine. It seems that when the program (Vegas) renders the Mpeg file with the audio file, it messes the dts audio file.
Can you pls give me some idea of how could I work around this? tks sfalcon |
27th November 2006, 19:28 | #30 | Link |
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I just burned a DVD and palyed it on a stand alone DVD player (Pioneer Elite). What I am trying to do is editing the audio of an mpeg 2 file (or just any DVD file), to convert it into a 5.1 wav file following Steve's guide and then put it back in a DVD disc. I succeeded to do so, but with wav files only. When I try to edit an mpeg 2 file, the rendering seems to mess it up... and the sound becomes a long hiss, instead.
thank you for yr trying to help. sds sfalcon |
27th November 2006, 20:55 | #31 | Link |
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Is your DVD player hooked up through a digital connection to your receiver (SPDIF or Toslink optical)? In order for the receiver's dts decoding to work, it needs to receive the signal digitally. If the player converts the signal to analog, all you'll get is loud hiss.
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28th November 2006, 01:10 | #32 | Link |
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Yes, my DVD player is attached to my AVReceiver thru a digital connection and also thru an HDMI link. I'm not sure, but I think that the hiss appeared after the mixed wav file was joined to the video part (rendering). If there is no rendering (like putting a background sound in a menu), the mixed wav file plays a beautiful DTS sound.
sfalcon |
28th November 2006, 17:36 | #34 | Link |
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By the way you said it, it seems to me that you have experience on this much. So, have you ever replaced the original sound of an MPEG 2 or VOB file with a DTS wav file delivered by SURCODE? If positive, did you play the DVD on a stand alone player and heard the DTS output normally?
This is exactly what I'm trying to do, but no success so far... tks sfalcon |
28th November 2006, 23:48 | #37 | Link | |
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Quote:
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29th November 2006, 06:06 | #38 | Link | |
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Quote:
Here's a workflow that will get you what you want:
NOTE: I've never done a DVD with DTSwave but I'm assuming that DVDLab will do the job for you. DVDLab supports DVD authoring with DTS streams. Hope this helps. DSP8000 |
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29th November 2006, 12:34 | #39 | Link |
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Fake DTS track on DVD
Hello guys!
Thank you for yr help! I found out what was wrong and now my DVD played with the DTS sound all right! Even using the Architect. The solution was to insert the wav file made by Surcode as a second audio, then Architect does not compress it and it plays fine tricking the AVReceiver decoder to think it's a real DTS source! tks a lot! sfalcon |
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