SACD - gapless playback

[Ghitulescu]

Originally Posted by Music Fan

The first thing to do is to copy the Sacd with a PS3 with Sacd Extract which makes an ISO.
http://code.google.com/p/sacd-ripper/downloads/list
Instructions ;
http://code.google.com/p/sacd-ripper...e/trunk/readme
Then you extract the dff files from the ISO with scarletbook.exe.
http://dl.dropbox.com/u/1370915/Audio/scarletbook.rar
Then you can convert Dsd to PCM (2.0 and 5.1) with several programs like Weiss Saracon, KORG AudioGate, Philips ProTECH Audio Format Converter, Pyramix Virtual Studio, maybe DiscWelder and maybe Foobar with foo_input_sacd.dll ;
http://sourceforge.net/projects/sacddecoder/files/
More informations there ;
http://www.stereo.net.au/forums/show...action-backups

I wonder whether there is a guide for gapless discs (live, electronic music etc.), sort of BIN+CUE....

[SeeMoreDigital]

Originally Posted by Ghitulescu

I wonder whether there is a guide for gapless discs (live, electronic music etc.), sort of BIN+CUE....

It's actually not too difficult to create your own .cue files

[pandy]

Btw - conversion from DSD to PCM is always lossy...

[Ghitulescu]

Originally Posted by SeeMoreDigital

It's actually not too difficult to create your own .cue files

The point was different.
Gapless discs, if compressed, will always be problematic to "rejoin", because of the buffer size differences (read padding) on both coding and decoding steps.

Yes, I know that DSD to PCM are lossy, however I am confident that the losses are not that great, in particular with titles that are available only as SACD (and are, according to Sony, processed in PCM then converted to DSD). On the other hand, HiRes FLACs or WAVs can be played on most of my players.

[SeeMoreDigital]

So you want to take individual tracks from different SACD/DSD sources and join them all together into one long track. And create a cue file that allows you to navigate to each individual track?

[pandy]

Originally Posted by Ghitulescu

The point was different.
Gapless discs, if compressed, will always be problematic to "rejoin", because of the buffer size differences (read padding) on both coding and decoding steps.

Well - real GAPless playback is Player functionality and it is not related to file format whatsoever...

Originally Posted by Ghitulescu

Yes, I know that DSD to PCM are lossy, however I am confident that the losses are not that great, in particular with titles that are available only as SACD (and are, according to Sony, processed in PCM then converted to DSD). On the other hand, HiRes FLACs or WAVs can be played on most of my players.

I would say that decimation filter reconstructing DSD (necessary step from DSD i.e. Pulse (1 bit) Density Modulation to PCM (multi bit) ) is directly responsible for sound characteristic (and as such it can be FIR or IIR and can have various unique flavors). Not sure if those filters (i mean software implementation) are "analog" or "digital" but i assume that from subjective perspective they can be perceived as "worse" or "better" (i.e. lossy or not).

[feisty2]

DSD...? never heard of it before, it's a commercial name for PDM according to Google
why would any one store audio in this format?

[SeeMoreDigital]

Originally Posted by feisty2

DSD...? never heard of it before, it's a commercial name for PDM according to Google
why would any one store audio in this format?

In this case, DSD stands for Direct Stream Digital. The audio stream format has been used on Super Audio Compact Disc's (SACD) since 1999 and is still being used...

Indeed, I 'pre-ordered' some new SACD disc releases yesterday

[Music Fan]

Originally Posted by Ghitulescu

The point was different.
Gapless discs, if compressed, will always be problematic to "rejoin", because of the buffer size differences (read padding) on both coding and decoding steps.

Interesting but succint, could you give more explanations about this ?
Are you sure gapless discs are problematic with the method I gave ?

[Ghitulescu]

Originally Posted by SeeMoreDigital

So you want to take individual tracks from different SACD/DSD sources and join them all together into one long track. And create a cue file that allows you to navigate to each individual track?

Yes, exactly.
But not exactly I want to have a huge file that should have cue points or indexes. I don't want to join files for the reason I already gave and is repeated below.

Originally Posted by pandy

Well - real GAPless playback is Player functionality and it is not related to file format whatsoever...

Well, even if a gapless-aware player can play the tracks with pause, if the tracks themselves are padded with NULL samples (to round the last encoding frame to its full size) the playback is still pausing (NULL samples are silence in audio). In short, like a file of 513 Bytes that has to be stored on two 512B sectors (while the rest of the 511 Bytes are padded with NULLs).
So while not technically an issue of the format, it still is related to the compression algorithm, and since a compression algorithm is related to a format (like MPEG to .MPG, MPEG1 layer 3 to .MP3 etc etc).

I might be worring too much, since I don't know the frame sizes for FLAC nor the indexing system of the SACD.

I hope it's clear now. If it's not possible, then it's Ok.

Thanks for input.

[SeeMoreDigital]

Ghitulescu,

As far as I'm aware Foobar offers a few DSD tools that might be able to help you. Not that I use it myself...

Sufficed to say, if you're project is related to any other audio formats types, then it's outside the scope of this topic.


Cheers

[pandy]

Originally Posted by Ghitulescu

Yes, exactly.
But not exactly I want to have a huge file that should have cue points or indexes. I don't want to join files for the reason I already gave and is repeated below.

Well, even if a gapless-aware player can play the tracks with no pause, if the tracks themselves are padded with NULL samples (to round the last encoding frame to its full size) the playback is still pausing (NULL samples are silence in audio). In short, like a file of 513 Bytes that has to be stored on two 512B sectors (while the rest of the 511 Bytes are padded with NULLs).
So while not technically an issue of the format, it still is related to the compression algorithm, and since a compression algorithm is related to a format (like MPEG to .MPG, MPEG1 layer 3 to .MP3 etc etc).

I might be worring too much, since I don't know the frame sizes for FLAC nor the indexing system of the SACD.

I hope it's clear now. If it's not possible, then it's Ok.

Thanks for input.

Well assume that payer need to be capable to decode 2 data streams at the same time and provide gapless fading between 2 data streams (trough fading/mixing) - so DSP need to be at least 2x times faster.

But perhaps my impression of gapless playback is different than Yours.

Sample padding/aligning seem to be less important.

[pandy]

Originally Posted by feisty2

DSD...? never heard of it before, it's a commercial name for PDM according to Google
why would any one store audio in this format?

Marketing... hype... audiophilism...

Concept of 1 bit delta sigma stream with strong noiseshaper sounds odd but lot of people believe that this format is superior to DVD Audio...

[Ghitulescu]

Originally Posted by pandy

Well assume that payer need to be capable to decode 2 data streams at the same time and provide gapless fading between 2 data streams (trough fading/mixing) - so DSP need to be at least 2x times faster.

But perhaps my impression of gapless playback is different than Yours.

Sample padding/aligning seem to be less important.

Actually not.
The player should work like Nero Burning Rom(e), using a buffer, and concatenate the files seamlessly.

Padding is very important.

[pandy]

Originally Posted by Ghitulescu

Actually not.
The player should work like Nero Burning Rom(e), using a buffer, and concatenate the files seamlessly.

Padding is very important.

In random mode you need to have capability to decode 2 streams and to perform fading between them - padding may help but real gapless playback assume that concatenation points (splices) overlap (at least partially). Buffering may work but can't provide random access and latency is unavoidable...

[Music Fan]

I'm not sure to understand what you mean ; is there a way (something to do at the authoring step or when burning the Sacd) to avoid the "pop" (or "glitch") sound that we hear between some tracks on some Sacd players ?
I have 3 Sacd players and the only one that does not have this problem is the Oppo. I found this explanation on Oppo's website concerning a firmware ;

-Some users have reported hearing pops or crackles between SACD tracks.
This issue happens to certain discs that have data frames that are not aligned with track boundaries. This version removes such pops or crackles.

What does exactly "track boundaries" mean ? Can we change it without conversion in pcm and re-make the Sacd (with Super Author for example) to avoid these "pop" sounds on all players and not only on Oppo (and maybe others) that solved the problem with a firmware ?

And I'm astonished you talk about Nero Burning Rom because I believed it was not Sacd compatible, Imgburn or UltraIso are generally used.

[Ghitulescu]

Originally Posted by Music Fan

What does exactly "track boundaries" mean ? Can we change it without conversion in pcm and re-make the Sacd (with Super Author for example) to avoid these "pop" sounds on all players and not only on Oppo (and maybe others) that solved the problem with a firmware ?

That's exactly what I meant.
Since DSD is a stream of bits, and SACD/DSD does not allot a fixed number of bits/bytes (it depends on the music) ie VBR, but the SACD sector is 2048 bytes, I am not sure how the track boundaries are set (they should be at sector boundaries - but this does not correspond to a fixed/foreseable place in the stream).

On the other hand, PCM has a fixed bitrate (CBR) and it's easier to "scrub"/cue to a position, knowing any of size and time.

Originally Posted by Music Fan

And I'm astonished you talk about Nero Burning Rom because I believed it was not Sacd compatible, Imgburn or UltraIso are generally used.

I was not talking about SACD in relationship with Nero, just mentioning Nero for its ability to create gapless tracks from separate files. I never used ImgBurn for audio.

[Ghitulescu]

Originally Posted by pandy

In random mode you need to have capability to decode 2 streams and to perform fading between them - padding may help but real gapless playback assume that concatenation points (splices) overlap (at least partially). Buffering may work but can't provide random access and latency is unavoidable...

I realised I was misunderstood.

I don't want padding.

I only explained why I don't want padding.

Gapless playback does not require two streams. It simply requires that once a stream ends, the next one should be in line to be processed. No player I know avoids a buffer - therefore indeed the content of two separate streams coexist temporarily, but the files themselves should have been processed one at a time.

This is how all hardware players (should) work. CD players work this way, if pure audio players (not MP3 players with computer drives)
Now, if the tracks/files are gapless, but the player pauses each time it loads the next one, this is IMHO due to two main causes
1. track boundaries (VBR) do not match sector boundaries - this way the player disregards the padded real bits and starts with the next track (which should have included the discharged bits)
2. track boundaries match sector boundaries but the player is programmed to read the tracks sequentially as tracks, and not to process the stream as a stream (ie ignoring indexes and stuff - but updating the information on display like TRACK 03 TIME 15:45.73)

I think the first case is the most common case.

[Ghitulescu]

Originally Posted by pandy

In random mode you need to have ...

Originally Posted by Ghitulescu

Gapless playback does not require two streams. It simply requires that once a stream ends, the next one should be in line to be processed. No player I know avoids a buffer - therefore indeed the content of two separate streams may and will coexist temporarily, but the files themselves should have been processed one at a time.

Live albums (or electronic music like JM Jarre, M Oldfield etc) are not made to be played in random mode, as they tell usually a story.
Yes, I know two (yes, only two) movies whose reels are designed to be played in a random order and still tell a coherent story. But these movies are the exception.

[Music Fan]

Originally Posted by Ghitulescu

1. track boundaries (VBR) do not match sector boundaries - this way the player disregards the padded real bits and starts with the next track (which should have included the discharged bits)

Ok, and do you believe there is anything feasible to make match track boundaries with sector boundaries, keeping Dsd as is ?
How to know if they match before exporting, burning, and listening Sacd ?
I'd like to avoid conversion in PCM, otherwise remake a Sacd would be a nonsense (after conversion, I would let PCM in PCM and make a dvd-audio).

And I wonder how did Oppo to remove these "pop" sounds, because I guess that if it can be done in real time by a standalone player, it can be done with some audio tools on a pc, but I may be wrong.