HELP! Mono dts-hd master! [Archive]

rik1138

13th August 2010, 01:54

You have access to the DTS Master Audio encoder? Try this:
Create a silent mono WAV file the same duration as the file you are trying to decode.
Use the DTS software to encode that WAV file into DTS-HD MA.
Copy the first 140 bytes of that new file, and paste them on to the beginning of the DTS file you are trying to decode (using a hex editor or something).

In theory, you should now be able to load the file into DTS Player and export the WAV files.
Removing this header information is a sort of copy protection implemented by DTS to prevent people from being able to decode the lossless format back into WAV/PCM files. I have no idea if creating a fake header like that will work, but I can't really think of a reason why it wouldn't... The header does have the duration in it, so it's possible if that's exact to the frame, it will give you some error... (Or, decode the core DTS to WAV, and encode that with the DTS software, that should give you a more accurate header to stick on the real lossless file.)

Worth a shot though, I suppose. I'm getting the same error with eac3to trying to decode a mono DTS-HD MA file.

ramicio

13th August 2010, 02:51

I just made a file out of the same spec and length wav. Eac3to won't even detect it. In a hex editor the header is wayy shorter, and has a crapload of empty space! Well the file I am trying to rip in the first place won't even open with the DTS-HD Master Suite tools, says it's DTS Express. This file is a huge mess! I'm beginning to think a receiver wouldn't even decode this stream as master!

Update: I extracted the core, and tried decoding it with the Arcsoft decoder, and it didn't like the core. Same error as the master file. I am on version 1.1.0.7. Is there a newer version out there I can try?

rik1138

13th August 2010, 07:37

Wait, hold on... A file encoded straight from the DTS-HD MA encoder will _not_ work with eac3to, the author of eac3to never programmed it to work with raw files, only files demuxed from a Blu-ray disc.

Apparently eac3to (or the ArcSoft decoder) can't handle mono DTS-HD MA. BUT, the DTS player _can_ play it back, and extract it to WAV files. But the player won't play a file ripped from the disc (as the authoring process strips off that 140 byte header).

So, what you need to do is create a file that the DTS Player can recognize (sounds like the DTS Player is your only option to get a lossless file).

So, take the first 140 bytes of the new file you just encoded, and paste them on the head of the file you ripped from the Blu-ray (thus, making that file 140 bytes larger in size).

Then, try playing that file (the original DTS with the new 140 bytes tacked on at the beginning) in the DTS Player software and see if it likes it. If it does, there's an option there somewhere for saving out the original WAV/PCM file...

It's a hack job, and I've never tried it, so it may not work... But thought I'd suggest since you obviously have access to the tools. This stripping of the header was done deliberately by DTS and the Blu-ray group for the sole purpose of making it difficult to restore lossless WAV files from the DTS-HD MA track. And, using the 'official' tools, it works. The DTS software will not recognize a ripped DTS-HD MA as lossless audio, that's by design.

ramicio

13th August 2010, 13:42

Libav handled the mono dts core just fine. Arcsoft couldn't even handle the core lossy mono dts. Total media theatre couldn't even play the sound when I played the disc! I don't even know how to copy and paste hex data of a certain range. The headers aren't even the same size. The studio did something creative to make this file. The suite software does recognize the file, as a DTS Express file. The bad file has a correct header that eac3to likes, the Arcsoft filter just doesn't like mono. It doesn't even like the header of an authored file, so I don't see how any of this will work to rip my audio.

rik1138

13th August 2010, 19:21

Everything you've said is expected. When you author a DTS-HD MA file unto a Blu-ray disc, the first 140 bytes of the file a deliberately removed. It's not possible to rip a Blu-ray disc and get the exact same file that use in authoring. It's part of the copy protection built into DTS. None of the DTS Master Audio suite programs will do anything with a DTS file ripped from a Blu-ray disc, they will always report incorrectly (as an Express file, or not even recognize it at all).

I assume you are using a hex editor to view the files? You should be able to select/copy the first 140 bytes of the file you encoded, and then paste them to the beginning of the file you ripped and save out a new file 140 bytes larger. I know the headers are a different size, that's what you have to fix if you want to try the DTS Player to decode the audio.

Apparently ArcSoft has a major bug in their DTS decoder that can't play back mono DTS files. That's going to prevent eac3to from working at all I assume...

How long is your audio file? I can try to create a dummy header for you that would be easier to attach if you aren't familiar with hex editing...

ramicio

13th August 2010, 20:13

It is 1:38:17. I have given up. NOTHING is going to decode this file's original lossless data to a wav file. The only thing that would work is to find a receiver with a decoder chip that outputs PCM and somehow capture that data by a computer. That is more trouble than this is worth. I'm not familiar with hex editing, and I cannot grasp the concept of hex when computers are really 0s and 1s. The only hope is that Arcsoft will fix their encoder, but I doubt they will because there are less than a handful of titles with mono audio tracks. I saw on their forum someone a while ago mentioned another title that was a mono DTS-HDMa that wouldn't decode, and there were 0 replies. It's pretty sad, because I thought just like THX, something has to be certified and to meet certain requirements to say they can decode something and use the DTS logo.

rik1138

13th August 2010, 21:42

BTW, what title is it?
You could always contact DTS and tell them how disappointed you are that they allow ArcSoft to put their logo on a product when it doesn't work. DTS would _not_ be happy to hear that their audio format is not working in a player, that looks bad for them. I have a couple of contacts there, I'll test the ArcSoft with some mono encodes and prod them with the results. :)

In the mean time, I'll try something that should be an easy test for you, don't delete the ripped DTS file yet!

ramicio

13th August 2010, 21:53

The m2ts file is going nowhere, I can always re-rip the audio. It is "National Lampoon's Vacation" that came out this year. The movie from the thread on the Arcsoft forum the other person had a problem with was "The Crazies" on blu-ray.

rik1138

13th August 2010, 21:59

Okay, try this:
Download the little attachment I've posted and unzip it.
Place it in the same fold as the DTS stream from the title you are working with
Open a command prompt, and navigate to the folder with the two files, and then type this:
copy header.dts+YourFile.dtshd NewFile.dtshd
Obviously 'YourFile.dtshd' should be the file name of the DTS file you ripped.
It will create a new file that should be exactly 140 bytes larger than the ripped DTS track. Now, see if the DTS Player application will recognize it.

I'm not sure it will work, but I figure it's worth a try...

http://forum.doom9.org/attachment.php?attachmentid=11386&d=1281733146

ramicio

13th August 2010, 22:03

I'm still wondering what this header has to do with anything? It will not help me decode this dts file into a wav. It will only let me play it. And the player that comes with the DTS Suite does in fact play the file, I think. I can't remember, I only did it once. I will have to check later today. I just can't perform any splicing or trimming functions on it.

rik1138

13th August 2010, 23:34

The header allows you to open it in the DTS Player software (and recognize that it's a DTS-HD MA mono lossless audio). The Player software has an option to save out the source file as .WAV

That's how you get your lossless WAV file, the DTS software itself can do it, but it needs a valid header on the file. DTS Master Audio is meant to be a studio archival format for storing lossless audio (as well as Blu-ray uses), so they also provide the tool to reverse the process.

And, with the header (if it works), you will be able to import the file into Stream Tools to edit/trim/whatever... It should recognize it as a DTS-HD MA lossless mono track. (Again, if it works...)

ramicio

13th August 2010, 23:36

Wow, what player is this? The player that came with the suite showed no such option. It seems like it's going to take forever for them to approve that attachment. Care to email it to me? timramich@gmail.com

OH MY GOD I found it. It's under the playlist!

rik1138

14th August 2010, 00:04

Try the link right above the attachment, I think that will work. I'll Email too, just in case.

Yeah, DTS provides handy tools, you just have to get their software to recognize the file! :)

Rik

ramicio

14th August 2010, 00:19

Well it worked, kind of. It only decodes 1:12:56 though.

ramicio

14th August 2010, 00:59

Well I finally got it! I tried decoding the core into a wav file just to make a DTS-HD Master file with the proper header for the time (it could be silence for all the encoder cares.) Well this software screws timings up big time. I only got 1:36:39 out of it on the PCM decode. So I just stretched the file to 1:50:00 and encoded that, then copied that header to the beginning of the ripped file, and ripped away. Success! I just figured it would either stop where it was supposed to or give an unexpected end of file error, but no error, it just stopped where it was supposed to. I think this thread could be informative to people in the future. Thank you very much, Rik! You are a genius!

BTW for you naysayers on quality...The new flac file is 62.6% its original size. The flac made from the lossy core (only 16-bit mind you) is 43.5%! Much more quality and actual sound information to fill a file with. If I was to take a studio music rip that is 24/192, they are usually around 40-50% when losslessly compressed. If I took that original PCM and resampled it down to 24/96 and compressed it losslessly, I would get a much higher number, usually 60-70%. This just shows that lossy is crap!

I am such a fanatic about audio quality that I do not or will not get a surround sound system as cool as it may be. I don't like the effect rooms acoustically apply to sound. For me it's more worthwhile to listen through headphones. What I do with all 5.1 Lossless tracks is split them up into wavs. Then in Goldwave I make a new blank wav file that is 24-bits/96khz. I mix the center and LFE into both channels, 0 dB. Then I select the left channel, and mix the front left and surround left into there, 0 dB, and I do the same for the right side. Of course with all those mixed in the levels are clipped, but Goldwave saves what you are editing in floating point :) So I just find the maximum peak, and take that much volume off minus 0.01 dB. Save, and compress with flac!

rik1138

14th August 2010, 01:10

rik1138

14th August 2010, 01:13

BTW for you naysayers on quality...The new flac file is 62.6% its original size. The flac made from the lossy core (only 16-bit mind you)

Ooo... I forgot about DTS being 24-bit... I was using a 16-bit WAV file... Maybe that's why there was a timecode problem? Not sure...

ramicio

14th August 2010, 01:21

Ooo... I forgot about DTS being 24-bit... I was using a 16-bit WAV file... Maybe that's why there was a timecode problem? Not sure...

That could be it! 16/24 = 2/3, and the ratio of the 2 different times comes out to be ~74%, so with the crazy messed up time codes, it was probably because of that!

nurbs

14th August 2010, 08:19

BTW for you naysayers on quality...The new flac file is 62.6% its original size. The flac made from the lossy core (only 16-bit mind you) is 43.5%! Much more quality and actual sound information to fill a file with. If I was to take a studio music rip that is 24/192, they are usually around 40-50% when losslessly compressed. If I took that original PCM and resampled it down to 24/96 and compressed it losslessly, I would get a much higher number, usually 60-70%. This just shows that lossy is crap!

Bigger doesn't mean better quality.
Suppose you have two flac files. File A was encoded straight from the source. File B was encoded after adding noise to the source. You'll want file B because it's bigger and you believe it to be better quality.

Difference in filesize after compression doesn't directly translate in audible difference.
See LossyWAV. (http://wiki.hydrogenaudio.org/index.php?title=Lossywav) Lossless compression will work much better after preprocessing with it, but it's transparent on most sources with the defaults, let alone higher quality settings.

ramicio

14th August 2010, 17:08

I was illustrating that the lower quality file (24/96) vs. the higher quality (24/192) has a higher percentage because the 24/192 does not use all of its dynamic range. When you resample it in half, it gets a more economical use of it when you losslessly compress it. And that in the case of my core vs. lossless files, the opposite is holding true, so therefore to the ears my lossless is much better. There were also obvious compression artifacts heard with the core file. When you losslessly compress something that has already been lossily compressed, it sees that it was compressed and it means it can compress lossless even more. I may not be wording it nicely, I'm not good at that.

Take a CD, rip a track to wav. Compress the wav with flac. Note the percentage. Compress the wav with a lossy format you feel is transparent. Take that lossily-compressed track and decompress it into a wav. Now compress that wav with flac, and note the percentage. It WILL be lower.

Take a bmp, rar or zip it. Compress the bmp to jpg. Save that jpg back to a bmp, compress that bmp with rar or zip. Yea, the bmp that came from the jpg should be smaller!

Also, with the noise debate, like it or not, vinyl has a lot of noise, but the fidelity is superior to CDs. Noise makes up part of the sound. They argue that there are frequencies on vinyl that are pointless because you can't hear them, but sound affects sound even if you can't hear it, so to me it is necessary to keep as much from the master recording intact as possible, even if its noisy. I was in my car at a red light, and a semi was sitting next to me. I had my windows partially open. I have a loud exhaust. Well the sound of his engine, which was very low in frequency, resonated with my exhaust note, and the way the windows were open, to make my car sound like it was idling half the speed it really was. I couldn't hear what sound from the truck was doing this. It was out of my audible range.

TinTime

14th August 2010, 17:26

Take a CD, rip a track to wav. Compress the wav with flac. Note the percentage. Compress the wav with a lossy format you feel is transparent. Take that lossily-compressed track and decompress it into a wav. Now compress that wav with flac, and note the percentage. It WILL be lower.

That depends on the precision you decode the lossy version to.

Thunderbolt8

14th August 2010, 17:55

Wait, hold on... A file encoded straight from the DTS-HD MA encoder will _not_ work with eac3to, the author of eac3to never programmed it to work with raw files, only files demuxed from a Blu-ray disc.
what you can do is demux the file DTS-HD MA file created by dts master suite with tsmuxer, then it becomes readable by eac3to. in a second step apply a delay of -21ms to that demuxed track so that the track will have exactly the same length as the original created file (seems to be a bug? of tsmuxer when demuxing such tracks)

ramicio

14th August 2010, 19:06

That depends on the precision you decode the lossy version to.

What precision? What does that even mean? You can't get back data from a lossy source because it tosses it out. If it didn't it would be called lossless. No matter how good your encoder and decoder is, you lose information. If you can't tell the difference between a lossy and a master, perhaps you have low end to average sound equipment, which would be most onboard sound out there. There are a few that feautre the x-fi, and are probably more onboard with good codecs behind them, but most are crap.

nurbs

14th August 2010, 19:40

And that in the case of my core vs. lossless files, the opposite is holding true, so therefore to the ears my lossless is much better. There were also obvious compression artifacts heard with the core file.
Maybe you can hear it. Maybe in this case the lossy track is worse than the lossless one, but considering it's a 756 kbps mono DTS track I doubt it. People often make claims like that, but when they actually do a blind ABX test they suddenly can't tell lossy from lossless any more.
I won't press you on your opinion, but you can't generalize that simply because it compresses better the quality will audibly worse. That's why I linked the LossyWAV page. Preprocessing with it can bring down the bitrate of the resulting flac by 20 to 40 percent and people are generally unable to ABX the difference, hence for most people on most samples it will be inaudible.

Also, with the noise debate, like it or not, vinyl has a lot of noise, but the fidelity is superior to CDs. Noise makes up part of the sound. They argue that there are frequencies on vinyl that are pointless because you can't hear them, but sound affects sound even if you can't hear it, so to me it is necessary to keep as much from the master recording intact as possible, even if its noisy.
I have several issues with that.
Random noise being introduced at playback doesn't make it higher fidelity, it makes it lower fidelity. I'm not saying that Vinyl is bad, I have several records myself and enjoy them. The noise adds a certain flair, but still fidelity is how well the input is reproduced and Vinyl is worse at that than CDs. The noise you hear when you play it back wasn't present in the master (mostly), it is an artifact introduced by the playback device.
There aren't any frequencies on Vinyl that can't be on CDs. On average a Vinyl record will be able to reproduce frequencies between 60 Hz and 18 kHz, maybe a bit more when it's in perfect condition. The dynamic range is about 12 bit under perfect conditions.
Basically that means everything that can be deliberately put on Vinyl, excluding noise added at playback, can be perfectly contained on a CD which supports frequencies up to 22 kHz with 16 bits dynamic range.

You can like the noise, you can argue that in some cases the Vinyl was made from a better master, but technically it's sound reproduction quality is inferior.

What precision? What does that even mean?

Precision refers to the bitdepth the samples will have after decoding. Due to the methods used in lossy compression you can decode the files to more or less arbitrary bitdepths. If you have an AAC file and you decode it to 16 bit PCM that's one representation of its content; by decoding it to 24 bit you'd get a more accurate representation.

TinTime

14th August 2010, 19:41

What precision? What does that even mean? You can't get back data from a lossy source because it tosses it out. If it didn't it would be called lossless. No matter how good your encoder and decoder is, you lose information. If you can't tell the difference between a lossy and a master, perhaps you have low end to average sound equipment, which would be most onboard sound out there. There are a few that feautre the x-fi, and are probably more onboard with good codecs behind them, but most are crap.

Fine. Why don't you run your proposed test? Let's assume that AC3 is our lossy codec of choice. Start with a CD track ripped to wav - orig_track.wav

Now run it through eac3to a couple of times to encode to AC3 and decode back to PCM:

eac3to.exe orig_track.wav inter_track.ac3
eac3to.exe inter_track.ac3 new_track.wav

Convert orig_track.wav and new_track.wav to FLAC. Which has the better compression ratio? Which is the higher quality?

ramicio

14th August 2010, 20:22

Basically that means everything that can be deliberately put on Vinyl, excluding noise added at playback, can be perfectly contained on a CD which supports frequencies up to 22 kHz with 16 bits dynamic range.

Not true at all. A sine wave is better represented in analog than digital, because it is pure. You can't make a CD rip sound better by upsampling it. It is what it is. If you record a vinyl at 24/44.1 and re-record it at 24/192, which will sound better? The 24/192. There is more horizontal resolution to accurately produce the wave at playback. For bits it would be the vertical resolution of the wave. Just because there is noise in vinyl doesn't mean you are throwing bits away, it is still higher fidelity and an actual analog waveform.

http://www.timramich.com/winewaves.png

Both waves are a 10 khz sine wave. Top file format is 24/192. bottom is 24/44.1. So you're saying a cd will sound better on a sole reason of having no noise? That's 10 khz. Less than half the total frequency a CD can reproduce. They become square waves at that point and sound bad. I know musical instruments don't produce their audible frequencies that high, except cymbals, which is the one thing that annoys me when listening to CDs. The cymbals sound artificial. CDs are fine for rap music or electronic stuff that is supposed to sound artificial. That's why I prefer to listen to my classic rock on 24/96 vinyl rips. Oh, and there's the turntable with lasers as a pickup. That would be the ultimate for someone who can afford it, as there would be no noise, and quite possibly the closest you can get to a master recording.

Ghitulescu

14th August 2010, 20:32

ramicio

14th August 2010, 20:48

This discussion is off-topic, however, I'll add something: one can't hear the digital sound: this will be converted to analog, and according to the D/A-converter in use, the "steps" will be attenuated.

Besides, the discussion vinyl vs. CD has its own topic. I own a medium price and quality LP-player, also a very good CD-player, medium class AVR and old-fashioned loudspeakers (not the type that has 2" tweeters and generates the basses through cunning reflexions) and the CD sounds better than the LP (on the same album), in every aspect, including the noise in the quit passages in classic music.

OK, so you have a crap turntable, and an awesome CD player, no wonder... Vinyl quality depends on the manufacturing. The first record I bought was Incredibad by The Lonely Island, brand new. The record came warped. Playback was even more disappointing. I also have some brand new records that are remasters of old rock. Boston by Boston sounds phenominal. Fool For the City by Foghat, too. Those are both very thick vinyl, and the person behind making it is making sure what you are paying extra for sounds awesome. My turntable is lower quality, too, and my cartridge isn't the best, but to me it sounds better than the CDs of the same album. Plus I enjoy my music with headphones :)

ramicio

14th August 2010, 21:03

Fine. Why don't you run your proposed test? Let's assume that AC3 is our lossy codec of choice.

OK...I have Hotel California by The Eagles, which the studio conveniently ripped to 24/192 directly from the master tape.

http://www.timramich.com/compression.png

1. the eagles - hotel california - 01 - hotel california 1.wav: This is the basis. I downsampled to 48 khz, kept it 24-bit.
2. the eagles - hotel california - 01 - hotel california 1.flac: This is the basis FLAC. 73.8%
3. the eagles - hotel california - 01 - hotel california 1.ac3: This is the first ac3 encode. 448 kbps, with eac3to.
4. the eagles - hotel california - 01 - hotel california 2.wav: This is decompressed from the eagles - hotel california - 01 - hotel california 1.ac3.
5. the eagles - hotel california - 01 - hotel california 2.flac: This would be the FLAC from the eagles - hotel california - 01 - hotel california 2.wav. 72.8%
6. the eagles - hotel california - 01 - hotel california 2.ac3: This is compressed from the eagles - hotel california - 01 - hotel california 2.wav @ 448 kbps.
7. the eagles - hotel california - 01 - hotel california 3.wav: Decompressed the eagles - hotel california - 01 - hotel california 2.ac3.
8. the eagles - hotel california - 01 - hotel california 3.flac: FLAC from the eagles - hotel california - 01 - hotel california 3.wav. 71.9%

This all is the reason why EXE files or JPG files don't compress much when you archive them in a ZIP or RAR. You get much better compression if you were to compress the source code of said EXE, or a BMP.

TinTime

14th August 2010, 21:11

Both waves are a 10 khz sine wave. Top file format is 24/192. bottom is 24/44.1. So you're saying a cd will sound better on a sole reason of having no noise? That's 10 khz. Less than half the total frequency a CD can reproduce. They become square waves at that point and sound bad.

This is absolutely not how digital audio works. The waves do not become square, they do not sound bad.

nurbs

14th August 2010, 21:15

A sine wave is better represented in analog than digital, because it is pure. You can't make a CD rip sound better by upsampling it.
I agree with both statements.

If you record a vinyl at 24/44.1 and re-record it at 24/192, which will sound better?
I'm not sure if you are talking about recording from a Vinyl (to PC) or recording what will eventually be put on a vinyl, I'm assuming the former.
They'll most likely both sound the same. Even if the first recording was made at 16 bit I'd still claim that. Higher bitdephts and samplingrates are beneficial for processing, so if you plan to do that recording higher can't hurt, but for listening it's a waste of space, especially if you record from a vinyl. Everything outside the bounds I've listed in my previous post is caused by dirt in the groves, wear of the groves and other imperfections of the playback system.
If you can actually hear a difference it's likely that something went wrong during recording. You could downsample and dither your high resolution track to 16 bit 48 kHz. An ABX test could then establish if there is an audible difference, assuming the playback hardware isn't to blame (I remember reading about one such case where the soundcard introduced artifacts at certain samplerates).
Even with samples specifically created for that purpose people are hardly able to ABX 16 bit from 24 bit and 48 kHz from 192 kHz under controlled conditions, so I don't buy that recording a Vinyl with a dynamic range and noise floor way above and beyond what the medium can provide will help.

Both waves are a 10 khz sine wave. Top file format is 24/192. bottom is 24/44.1. So you're saying a cd will sound better on a sole reason of having no noise?
What I'm saying is that if you take the same master and use it to produce a Vinyl and a CD the CD will be a more accurate representation of that master, because it has a higher frequency range, higher dynamic range, lower noise floor and playing it back lacks the artifacts that are introduced when playing back vinyl.
Let's say you actually take your 10 kHz sine wave and put it on Vinyl and CD. You rip both, the former you record to 24 bit 192 kHz and the latter you upsample to that (or you downsample the former, doesn't matter). If you now compare both recordings to your master the CD will be closer to it because it didn't have a boatload of artifacts introduced during pressing and reading. Vinyl is analog, but the groves are neither pressed nor read with infinite precision. The values I've given are real live limits verified through extensive testing.

Again, I'm not saying you shouldn't enjoy Vinyl. All I'm saying is that when it comes to fidelity it is inferior to CD.

I recommend you go over to Hydrogenaudio (http://www.hydrogenaudio.org/forums/index.php?) and take a look at their wiki pages about Vinyl (http://wiki.hydrogenaudio.org/index.php?title=Category:Vinyl), especially the FAQ (http://wiki.hydrogenaudio.org/index.php?title=FAQs_(Vinyl)) and the Myths (http://wiki.hydrogenaudio.org/index.php?title=Myths_(Vinyl)) section.
By the way if you actually have samples where the difference between 16 bit 48 kHz and 24 bit 192 kHz is audible and you can reliably identify them by ABX tests the people there will be very interested in you and your samples.

ramicio

14th August 2010, 21:21

This is absolutely not how digital audio works. The waves do not become square, they do not sound bad.

But when you put a whole ensemble of sounds together and compress that waveform down to something like a 44.1 khz CD you start missing information. I will agree if you will agree that making something digital you lose information. My argument is just the quality of the sound you get in the end. I can totally agree with these theories, but when it comes to being in the real world, I hear much more detail of voices and instruments from the same vinyl album vs. CD.

This started as a help thread because I wanted the lossless. I wasn't asking for anyone's opinion on if the lossy would be a neglible difference. Rik was helpful, and he didn't present any opinion on why I wanted the lossless. For that I thank him. I'm not telling anyone they should buy a huge hard drive and have an entire music collection in lossless, as I do. If you like what you have and like to listen to it, great! For me I enjoy the best quality I can get. For video, I don't care as much, as long as I don't see compression artifacts I'm satisfied. Transcoding video takes a long time, this is why I don't care. Audio is different story and to transcode things takes me little time. I myself can tell a huge difference between DVD-Audio @ 24/192 and the same album on CD.

TinTime

14th August 2010, 21:55

OK...I have Hotel California by The Eagles, which the studio conveniently ripped to 24/192 directly from the master tape.

You've changed your test parameters (this is not CD audio) so you've kind of defeated the point I was trying to make. My point is that the compressibility of audio decoded from a lossy encode will be dependent on the precision (bit-depth) that the lossy audio is decoded to so it's not a particularly useful way to measure audio quality.

But when you put a whole ensemble of sounds together and compress that waveform down to something like a 44.1 khz CD you start missing information. I will agree if you will agree that making something digital you lose information. My argument is just the quality of the sound you get in the end. I can totally agree with these theories, but when it comes to being in the real world, I hear much more detail of voices and instruments from the same vinyl album vs. CD.

Yes. You lose frequencies above whatever the low pass filter is set to going in and out of digital, and you introduce noise in the form of dithering.

I'm not arguing that you should prefer CD to vinyl. That's just personal preference. I'm with you - I love records. I was just saying that your graphs with their stair steps are misleading.

This started as a help thread because I wanted the lossless. I wasn't asking for anyone's opinion on if the lossy would be a neglible difference. Rik was helpful, and he didn't present any opinion on why I wanted the lossless. For that I thank him.

Again I never offered an opinion on lossy vs lossless. That's for you to decide. I was just suggesting that decoding a lossy track and then seeing how it compresses with FLAC isn't necessarily a useful metric to judge audio quality because other factors can come into play.

ramicio

14th August 2010, 22:07

Dude, it's called lossy for a reason. You CANNOT keep information on lossy compression. This is why it's called LOSSY. It doesn't matter what sound format I used for this. To encode a 16 bit file into ac3 and have it decode to a 24 bit just shows there is loss and it needs the extra info of 24 bits to give you the info instead of rounding it. Even at 24 bits you still lost info. If you downconvert to 16 bits it will just round off the numbers. You weren't the one to tell me to just grab the lossy, with no help, it was in post #4. For me, lossy compression is fine for my iPhone, where there is not a lot of space, but for my pc, its lossless. I really hate mp3, even at 320 it sounds bad. I like AAC. At 320 it sounds damn good, and way better than mp3@320. I don't need crazy quality for my phone, I have nice Bose earbuds, but earbuds are limited in quality, and whenever I listen to music with it, the environment isn't quiet. The thing I really don't like about CD is the way they compress it. If you look at the same song on CD, vs a captured vinyl, or if you're lucky a DVD-Audio, like every sound on the CD will be at peak volume. I have witnessed a few CDs that do sound awesome, and their waveform is quieter and more natural.

nurbs

14th August 2010, 22:49

The thing I really don't like about CD is the way they compress it. If you look at the same song on CD, vs a captured vinyl, or if you're lucky a DVD-Audio, like every sound on the CD will be at peak volume.
That's done in mastering and is a result of the Loudness War (http://en.wikipedia.org/wiki/Loudness_war). It's not a limitation of the medium. DVD-Audio offers a higher possible dynamic range than CDs, while Vinyl has a lower dynamic range than both.

ramicio

14th August 2010, 23:44

That's done in mastering and is a result of the Loudness War (http://en.wikipedia.org/wiki/Loudness_war). It's not a limitation of the medium. DVD-Audio offers a higher possible dynamic range than CDs, while Vinyl has a lower dynamic range than both.

But lower dynamic range doesn't make vinyl sound bad, or it would sound worse, and it sounds better. You're only looking at one area of sound, volume. The signal is not messed with and the dynamic range is not compressed like CDs. It is the authoring, but there is a lot more care taken when the put something on vinyl vs. CD. What they're doing to CDs today makes them sound like crap, and their vinyl counterpart WILL sound better.

rik1138

15th August 2010, 01:01

what you can do is demux the file DTS-HD MA file created by dts master suite with tsmuxer, then it becomes readable by eac3to. in a second step apply a delay of -21ms to that demuxed track so that the track will have exactly the same length as the original created file (seems to be a bug? of tsmuxer when demuxing such tracks)

Heh, this thread derailed fast... :cool: I'm glad we solved the problem FIRST, the thread can do what it wants from here on out I suppose.

Just wanted to comment on this, Yeah, I have done that before. To make eac3to recognize an original DTS file, you have to remove that 140 byte header. I just delete it with a hex editor, but muxing/demuxing it with most of the tools should achieve the same result.

sub24ox7

15th August 2010, 07:33

xkodi

15th August 2010, 11:21

unfortunately after doing more tests this approach doesn't seem to work well when you use "fake" header, even if that header is the same as the one of the original stream, what i mean:

1. one 96kHz 24-bit 7.1 channel DTS-HD MA stream from Blu-ray is:

1.1. demuxed with eac3to to file.dtshd, i.e. with missing 140 bytes header
1.2. decoded with eac3to to eight 96kHz 24-bit 7.1 mono WAVs (it's confirmed that those WAVs are bit-perfect decode, because the same disc has the track in LPCM and TrueHD, which can be decoded bit-perfect and all 3 decoded streams to WAVs are byte-by-byte identical)

2. from 1.2. encode DTS-HD MA stream with exactly the same parameters as the original DTS-HD MA stream from the Blu-ray

2.1. use the 140 bytes header from 2. and merge it with 1.1.

3. decode 2 and 2.1 using the player:

3.1. first about 10-11MB of 2.1 are decoded bit-perfect and byte-by-byte identical to decoded 2., but then the rest of decoded 2.1 from the player is not correct

3.2. decoded 2. is byte-by-byte identical to 1.2., which is correct result

4. the explanation why doing 2.1. is not enough to decode the stream correctly: yes adding the header allows you to load the stream in the player and start decoding it, but comparing 1.1. and 2. shows that DTS-HD MA encoder adds not only 140 bytes header to the stream, but some tail at the end of the stream with unknown length, which apparently is used by the player during the decoding, because removing that tail results in inconsistent result in 3.

i don't know if i explained it clear enough, but in short at least what my tests show is that adding just a 140 bytes header results in stream that is not correctly decoded by the player and the tail is also required, but creating correct tail for stream that is missing the tail, which will allow correct decoding by the player, seems close to impossible at least at the moment.

maybe, someone can run similar tests to mine above and find something else useful.

sub24ox7

15th August 2010, 11:34

nurbs

15th August 2010, 12:21

But lower dynamic range doesn't make vinyl sound bad, or it would sound worse, and it sounds better. You're only looking at one area of sound, volume. The signal is not messed with and the dynamic range is not compressed like CDs. It is the authoring, but there is a lot more care taken when the put something on vinyl vs. CD. What they're doing to CDs today makes them sound like crap, and their vinyl counterpart WILL sound better.
If you use a different master for Vinyl it can sound better if that master wasn't DRCed to achieve a higher volume as is usually done these days. For most stuff mastered in the last 20 years you'll have the same master used for both CD and Vinyl. There are exceptions of course, but mastering isn't cheap and demand for Vinyl is low to begin with. What usually happens is that the same master is used, but the recording level is lowered and treble limiting (essentially lowpass filtering) is applied before being pressed on Vinyl. That a special, better master has to be created, or usually is created for pressing on vinyl is one of the many myths Audiophiles like you believe in.

madshi

15th August 2010, 12:40

Can anybody upload a small DTS-HD Master Audio Mono sample for me?

Thunderbolt8

15th August 2010, 12:48

dont have any dtsma 1.0 track

xkodi

15th August 2010, 12:52

well this is indeed not good news at all :( since I just got the ading the 140byte header to work, I wonder if running through any of the DTS-HD stream tools would add the tail EDIT: I entered a dtshd track into the stream tools file info and it says that the timecode chunk is missing.

if i look at the tail there is section inside about the time-code, but i guess it contains some other information too, which prevents the player to parse the stream correctly without the tail, because at least i don't have any other explanation why at the beginning the decoding is good and correct and then after first several bytes it's not good anymore.

sub24ox7

15th August 2010, 17:19

Madshi I can upload a small sample, how big?

madshi

15th August 2010, 17:30

10MB would be nice, if there's any sound in the first 10MB. If it's all silent, then a bit more.

sub24ox7

15th August 2010, 17:50

ok its 19 MB http://www.mediafire.com/?tduqu4qnb24ugzb

madshi

15th August 2010, 17:56

Thanks.

sub24ox7

15th August 2010, 17:58

np I hope you can get this sorted out as I know if there is anyone who can its you.

madshi

15th August 2010, 18:58

np I hope you can get this sorted out
It seems I can... ;)

sub24ox7

15th August 2010, 19:20

Wonderful news!!!!

madshi

15th August 2010, 20:11

http://forum.doom9.org/showpost.php?p=1426340&postcount=10431

sub24ox7

15th August 2010, 21:57

WOW!!!!! that was fast!!!!!! and works perfectly!!!!
http://i307.photobucket.com/albums/nn293/emonizer/em1/headbang2.gif (http://emoticonizer.info)

ramicio

16th August 2010, 01:20

can someone please post a 24bit header or telll me how to make a 24bit header so I don't have to stretch the file or tell my how to stretch the file as I have the the national Lampoons european vacation disc with mono DTS-HD MA also this method" Open a command prompt, and navigate to the folder with the two files, and then type this:
copy header.dts+YourFile.dtshd NewFile.dtshd" results in a 265KBfile so it does not work :(

you gotta do:

copy /B header+original.dtshd newfile.dtshd

The /B switch denotes binary. But now the mono decoding in eac3to is fixed! Something Arcsoft wouldn't fix for play their player.

Ghitulescu

16th August 2010, 19:51

OK, so you have a crap turntable, and an awesome CD player, no wonder... Vinyl quality depends on the manufacturing. The first record I bought was Incredibad by The Lonely Island, brand new. The record came warped. Playback was even more disappointing. I also have some brand new records that are remasters of old rock. Boston by Boston sounds phenominal. Fool For the City by Foghat, too. Those are both very thick vinyl, and the person behind making it is making sure what you are paying extra for sounds awesome. My turntable is lower quality, too, and my cartridge isn't the best, but to me it sounds better than the CDs of the same album. Plus I enjoy my music with headphones :)
I have an above average turntable, I was referring to a turntable that doesn't cost 15000€, only a fraction thereof, but yields a reasonable -78 dB own noise (after EQ) with an MC cartridge and low flutter, that's 20 dB more than any LP can give - believe me, it's just placebo, you simply can't obtain more, unless one buys the "heavy" LPs which bring 5 dB more. The reason I keep it is because I own some plates that are not yet on the market and probably never will.

I understand that the CDs are bad masterized today, I've said that long time ago, but I still prefer the CD over the LP for most of my music. I can live with the distortions in the high frequency range as long as no low frequency noise is present (you simply cannot avoid this on a LP, due to the RIAA EQ-curve) - you may check my [older] posts concerning the low noise effects on the brain.

The record industry is pushing again the LP as the LP cannot be 100% copied, but they cannot fool me, I'm in this business for long.
=============================
PS: using headphones, that are generally not very well suited for infrasounds, partially avoids the negative effects associated with the basses.

PS2: the only enemy of the CD sound is the player, most of them are simply out of specs (in terms of jitter). Once one manages to get rid thereof, then the CD would sound almost perfect.

PS3: anything more than 44.1/48 kHz and 16b is for LP an overkill. There are no significant frequency over 16 kHz, there is noway a noise floor beyond 60 dB (98 for CD).

ramicio

16th August 2010, 20:05

OK, but vinyl still sounds better. There is more depth to the sound there! At least to my ears. I am not making this up. If you rip a cd to a computer you are getting raw data, it either works or it doesn't. If you miss a bit from the stream it will mess up the rest. So playing a ripped CD in flac would be a bit-perfect representation of that. Still doesn't sound as good as my ripped vinyl. If I resample a vinyl rip down from 24/96 to redbook quality, it WILL NOT sound as good. As I said there are spots in the waveform which I believe will just be ignored when downsampling. The noise you hear "goes away" once the music actually plays. The music industry didn't push vinyl back in. People want it because it obviously sounds better. And it can't be for DJing either because they have digital turntables now for scratching and all that stupid stuff.

TDiTP_

28th December 2010, 04:43

xkodi
http://forum.doom9.org/showthread.php?p=1426223#post1426223

I can't confirm it.
i did all that you but in my case decoded 2.1=decoded 2 (byte-in-byte), both identical to 1.2. But i added 2 frames (dts-hd) delay to 2.1, i.e. i added 21 ms of silence to decoded 2.1. Then ok.
My "dts-hd ma 5.1 48/16 core: 768kbps" was from this Blu-ray: http://www.blu-ray.com/movies/9-Songs-Blu-ray/5789/

comparing 1.1. and 2. shows that DTS-HD MA encoder adds not only 140 bytes header to the stream, but some tail at the end of the stream with unknown length

yes, but in my case these "some tail at the end of the stream" didn't prevent.

What do you think?