Converting DSD Files to FLAC

[ssully]

True of coarse, that's why they made the setting adjustable. I've yet to have a problem with using the +6 dB setting however. If clipping does occur you can simply reconvert at the lower setting. It would be useful if there was a listing of those "hot" releases.

That's a great suggestion. I wish I'd kept all the logs.

 

[ssully]

I didn't even know you could apply replay gain tags to DD and DTS files. Good to know.

It's not a matter of adding tags (the only way I know to do that is to save raw DD or DTS files in a special format --SPDIFWav -- that allows FLAC conversion-->tagging. Audiomuxer can do this.*)

It's a matter of having replaygain -- or any DSP -- 'on' in your foobar2K player. If any DSP is active in foobar, the digital output is not bit perfect. DD and DTS (raw) streams have to be a bit perfect for the AVR to decode it. Otherwise, white noise. (This is also why the foobar volume has to at max whenever streaming a DD or DTS file)


*Nowadays since software decoders are excellent and work with DTS 96/24, HD, etc, it's just easier to rip a DD or DTS to WAV-->FLAC and be done with it. But I haven't had to deal with Atmos...yet.)

 

[ssully]

I don't think it has been mentioned on this thread but if you have a Mac, I highly recommend DSDMaster (in the App Store) for converting DSD files to PCM. It has a feature to automatically normalize over file groups which take the guesswork out of the whole +3/+6 gain part of the equation. It will analyze all the tracks on an album and normalize them all the same amount without any going over.

https://apps.apple.com/us/app/dsd-master/id829431988?mt=12

If it does this on a per track basis, it changes one aspect of the original album mastering-- the peak differences between tracks on the album. Most classic albums don't reach the same peak on every track. That matters if you like to play whole albums and hear them 'as intended'.

 

[JediJoker]

If it does this on a per track basis, it changes one aspect of the original album mastering-- the peak differences between tracks on the album. Most classic albums don't reach the same peak on every track. That matters if you like to play whole albums and hear them 'as intended'.

The way @Cheezmo worded it, it sounds as if it will do per-album normalization, whereby it finds the track with the highest peak and normalizes all tracks by the same amount referenced to said peak.

 

[LuvMyQuad]

It's not a matter of adding tags (the only way I know to do that is to save raw DD or DTS files in a special format --SPDIFWav -- that allows FLAC conversion-->tagging. Audiomuxer can do this.*)

It's a matter of having replaygain -- or any DSP -- 'on' in your foobar2K player. If any DSP is active in foobar, the digital output is not bit perfect. DD and DTS (raw) streams have to be a bit perfect for the AVR to decode it. Otherwise, white noise. (This is also why the foobar volume has to at max whenever streaming a DD or DTS file)


*Nowadays since software decoders are excellent and work with DTS 96/24, HD, etc, it's just easier to rip a DD or DTS to WAV-->FLAC and be done with it. But I haven't had to deal with Atmos...yet.)

I do use Replay Gain, but then again, I rip everything to FLAC. I fully understand about the white noise thing with DTS.

The only way I know of to rip Atmos is the lossy M4A format or MKV. With M4A, I'm just not quite sure how lossy it truly is. When I play back the resulting files they sound fine. MKV does as well (the initial rip is to MKV, followed by a conversion from MKV to M4A). I don't have a good way to properly tag MKV for audio though. I can tag the M4A files, but It doesn't seem like Replay Gain works with them. I hope there is a new generation of ripping/tagging software just around the corner to remedy all this.

 

[LuvMyQuad]

The way @Cheezmo worded it, it sounds as if it will do per-album normalization, whereby it finds the track with the highest peak and normalizes all tracks by the same amount referenced to said peak.

That's how Replay Gain tags work as well. It gives a choice on what peak levels are being used.

 

[JediJoker]

That's how Replay Gain tags work as well. It gives a choice on what peak levels are being used.

Right, but that's on playback, at the software player level. DSD Master will apply gain to the audio files themselves in order to bring them closer to full-scale in PCM. It serves a different purpose.

 

[himey]

That's how Replay Gain tags work as well. It gives a choice on what peak levels are being used.

The difference with replay gain is you can turn it off, unlike what is being discussed here, which you probably already know.

Edit...some joker beat me to it

 

[Cheezmo]

If it does this on a per track basis, it changes one aspect of the original album mastering-- the peak differences between tracks on the album. Most classic albums don't reach the same peak on every track. That matters if you like to play whole albums and hear them 'as intended'.

It analyzes each track, but normalizes the entire "batch" so that no track exceeds the peak level but all tracks have the same normalization. That is the whole point to maintain the relative volume of all the tracks on an album.

 

[ar surround]

First, I use Sonore iso2dsd (a free Javascript app) to rip a SACD iso image to DSD files.
(I feel like there should be a better way to do this... A Java app? ... But it appears to work and it's simple.)

I'll use XLD to convert to PCM and I save that in FLAC in the end.
Settings here are important!
You want 32:1 decimation to convert the DSD losslessly to PCM. This results in a sample rate of 88.2k.
Anything else gives a lossy conversion and the lossiness goes way beyond normal PCM to PCM sample rate conversion!

The levels on DSD program can end up anywhere from 6db low to 6db over zero after conversion to PCM.
If you have a hot example and convert it straight to FLAC (or any format in 24 bit), you will clip! Conversely, a lower level example will lose precision (even if only 1 bit). So it is critical to convert to floating point first!

Convert the DSD to 32 bit floating point wav using 32:1 decimation.
Open the wav files in your favorite DAW app.
Normalize the levels to just below zero.
Now render those files to 24 bit fixed and convert to FLAC.

Re: 176.4k sample rate sounding better than 88.2k.
You may have hardware converters that perform better as ultra HD sample rates vs. HD sample rates. (In the same way that most consumer converters found in AVRs run better at HD than SD.)
You could convert some program in 88.2k to 176.4k and A/B listen to try to confirm that. There's no actual meaningful audio content captured above 20kHz. HD sample rates are all about the machines we build (AD & DA converters in this case) running better. I'll assume you were being critical and this wasn't a red hearing chasing a different volume level. (You must match levels within 0.1db or you will pick the louder program as sounding better.)

Is there an easier way to go through all of this on a Mac? Once a file gets converted to WAV, all of the tagging disappears.

 

[Marplot]

Sox going straight from dsd to flac while retaining tags.

https://sox.sourceforge.net/

 

[Cheezmo]

Is there an easier way to go through all of this on a Mac? Once a file gets converted to WAV, all of the tagging disappears.

I use DSD Master (in the App Store). It handles all the tricky normalization (determining the level for an entire album and adjusting all the tracks the same to avoid clipping.

Just drag the DFF files, convert to FLAC. Easy peasy. Handles 5.1 just fine.

Only downside is that it isn't free and hasn't been maintained in a while but it works find on my M1 MacBook Pro.

 

[himey]

They are definitely different. L/R/C shown.

What did you use to convert the SACD to PCM so you could post the waveform? Did you bump it down 6db?

 

[pat bateman]

What did you use to convert the SACD to PCM so you could post the waveform? Did you bump it down 6db?

Foobar has a SACD decoder. There are not any configurable options however, so I'm not sure if its doing any db adjustments during the process.

 

[sjcorne]

There are not any configurable options however, so I'm not sure if its doing any db adjustments during the process.

You have to go to file -> preferences -> SACD to make the level adjustments. Most rips come out way too quiet without doing this, but some are okay.

 

[JediJoker]

You have to go to file -> preferences -> SACD to make the level adjustments. Most rips come out way too quiet without doing this, but some are okay.

View attachment 87280

I don't recommend a blanket +6dB when converting to PCM. For many SACDs, that will produce clipping.