SONAR's resampling DSP was built in-house and has been time tested over many versions. As I said its very accurate and doesn't introduce any audible artifacts let alone gain changes. There's no need for "options". It just works transparently when resampling is needed.

lfm
My concern has been since I noticed that default I had in Reaper made a 44k original reference track of a professional recording was resampled running the project in 48 - and it created peaks 1.6dB higher, actually overs then on a pro recording.

The 1.6dB "higher" peaks were almost certainly just intersample peaks that were already present in the 44.1kHz recording. It's just that the SRC moves where the samples are on the waveform.
 
It's not at all unusual for an SRC to cause you to see peaks between +1dB up to about +3dB compared to the original, but these peaks were present in the original - they just happened to fall between the existing samples.

Anderton
If you look at the test results for Reaper referenced in post #2, you'll see that Reaper's algorithms aren't particularly good and certainly don't stack up to SONAR, Logic Pro X, etc. I don't know if they've improved since those tests were made, but if not, it's not surprising you wouldn't find them transparent. Really, the only reason to offer a choice of multiple algorithms is if there isn't one good one. That way you can choose the one that's "less bad."

Despite the really ugly looking stopband ripple on the Reaper SRC's, except for the really pathetic (!!!) Reaper 2 IIRx2 SRC, I wouldn't expect them to be a problem except when there was very high level, very high frequency content present in a signal.

Noel Borthwick [Cakewalk]
SONAR's resampling DSP was built in-house and has been time tested over many versions. As I said its very accurate and doesn't introduce any audible artifacts let alone gain changes. There's no need for "options". It just works transparently when resampling is needed.

Really good to hear.
 
Sonar import was actually among those I never saw do ampltiude changes on importing cd reference material - as a reference talking to Cockos about it. I didn't know for sure if you just normalized something in the end - so could not tell if algo was better or not, looking at amplitude alone. Cockos team just claimed - this is to be expected resampling, peaks will be different.
 
I remember on of the last updates that Vsampler did(10 years ago, known to Sonarites) was adding a range of new resampling algos.
 
Article linked to in OP, also states there are differences and one should really convert before using the crappy stuff in daws.
 
Where you really have to make compromises is where you do things in realtime, I guess. And you have a better/heavier one for rendering than realtime playback. Also what they do in Reaper as well as Vsampler.

lfm
Article linked to in OP, also states there are differences and one should really convert before using the crappy stuff in daws.

I wouldn't put too much stock in that article as there are a number of technical errors in it.

Anderton
Really, the only reason to offer a choice of multiple algorithms is if there isn't one good one. That way you can choose the one that's "less bad."

When was reality that simple - nothing I ever saw.
 
Most obvious is realtime stuff that may be too heavy on cpu.
That pretty much what I discovered using Ozone 2 and 3 ten years ago - still sounding digital.
 
Building math for good algos is what you can base a company on - look at Softube.
 
- What we provide is so good that you don't need anything else
... that is a bit too much sales talk

drewfx1
The 1.6dB "higher" peaks were almost certainly just intersample peaks that were already present in the 44.1kHz recording. It's just that the SRC moves where the samples are on the waveform.
 
It's not at all unusual for an SRC to cause you to see peaks between +1dB up to about +3dB compared to the original, but these peaks were present in the original - they just happened to fall between the existing samples.

..and if you use another algo that intersample peak is not there - so which algo is doing it right?
 
I chose a more heavy algo, same type, in Reaper and 1.6 went down to 0.6 dB - still claiming nothing to do with algos?
Only source material?
 
As a reference I did imports to Sonar with no such overs - but did not know at the time if Sonar possibly did some normalizing afterwords. Noel has clarified algo is not doing that.
 
1.6dB - that is 20% - you get intersample peaks that size?
I'm not buying that explanation - not of that size.
 
And if it's there it shows on analog signal later as well.
I haven't tested this case at the time on analog side, but professional equipment would, I guess.
 
 

lfm
- What we provide is so good that you don't need anything else
... that is a bit too much sales talk

Well, think about it for a second. If what goes into the algorithm sounds exactly the same as what comes out, I don't know why you'd need anything else...but if it doesn't sound the same, then you DO need something else!
 
This is also a good time to mention SSL's X-ISM free metering plug-in, which indicates if there's inter-sample distortion that conventional meters don't show.
 
 

lfm

drewfx1
The 1.6dB "higher" peaks were almost certainly just intersample peaks that were already present in the 44.1kHz recording. It's just that the SRC moves where the samples are on the waveform.
 
It's not at all unusual for an SRC to cause you to see peaks between +1dB up to about +3dB compared to the original, but these peaks were present in the original - they just happened to fall between the existing samples.

..and if you use another algo that intersample peak is not there - so which algo is doing it right?
 
I chose a more heavy algo, same type, in Reaper and 1.6 went down to 0.6 dB - still claiming nothing to do with algos?
Only source material?

 
Different algorithms can have different amounts of high frequency attenuation, phase shift, aliasing, etc. You can't determine what causes a level change without examining exactly what changed.
 
I'm assuming you're talking about peak levels, where differences would not be surprising. Changes in RMS would indicate the SRC was changing the level.
 

1.6dB - that is 20% - you get intersample peaks that size?
I'm not buying that explanation - not of that size.

 
You can easily get intersample peaks of that size or larger, depending on the audio itself, and, for maximum intersample peaks, over how long a period you measure. 
 

 
And if it's there it shows on analog signal later as well.
I haven't tested this case at the time on analog side, but professional equipment would, I guess.

 
Yes. In theory, a DAC should allow for a few dB of headroom above 0dBFS to allow for intersample peaks. In practice that might or might not be the case. 

Anderton

lfm
- What we provide is so good that you don't need anything else
... that is a bit too much sales talk

Well, think about it for a second. If what goes into the algorithm sounds exactly the same as what comes out, I don't know why you'd need anything else...but if it doesn't sound the same, then you DO need something else!
 
This is also a good time to mention SSL's X-ISM free metering plug-in, which indicates if there's inter-sample distortion that conventional meters don't show.

 
I won't argue with "sounds exactly the same" - but maybe consider that's theory, not real life.
 
And think about this:
One track conversion - let's say moving a project into a daw that will convert things.
When doing listening, I usually spend an hour back and forth to even conclude what result is, comparing. Differences are subtle.
Let's assume one track is inaudible to hear differences.
Import 20 tracks and listen to sum of these on master - then there might be even disturbing artifacts as a total. And make it 40-50 tracks and it's even more obvious.
 
I read an article on preamps, why studios spend $3000 on a preamp when most of us feel that $1000 preamps sound as good as anything.
But this tiny extra bit of clarity on each recorded track makes quite a difference when coming to the total mix of it all.
 
So why do studio recordings usually sound just that little bit better than your own stuff - look at all parameters. Such cumulative things may be part of reason.
 
I read loads of posts over creating "air", and do eq, saturation stuff and I don't know what - trying to compensate for less good recordings. Conversion stuff may be part of it.
 
I recently learned about resampling in Superior Drummer, I had an instability issue in 2.4.1, that was fixed now in 2.4.2. It only occured if not having project sample rate to 44k, since it's recorded at 44k. So that raised my concern again about resampling issues - if to maybe go back to 44k. Have to do some tests on this before deciding. I do maybe half of tracks as recorded analog stuff and half ITB.
 
I'm just trying to learn my tools, and what is the best way to use them - not to create problems showing up in final mix.
 
I bought this album that both had cd and some 24/96 master on a companion dvd.
- This will be fun to see if there really is a noticeable difference
I thought.
So I fixed up some strap in my external DAC to sense 96k and went on with testing this.
a) first I found that dvd player is restricted by industry to downsample to 48k - that was a bummer.
b) so I went on looking for computer based players that do 96k, and do asio etc so my RME card get the proper data. This just to find that the supplied 24/96 analogue master they supplied sounded crap and dull as anything. So why they do this is mystery to me - it's supposed to be a bonus, isn't it bringing something extra to you?
 
So you live and learn, don't you.
 
Thanks for the SSL tip, really interesting.
I guess if doing x8 upsampling(with a good algo) or something within a plugin, you will also get digital values for hidden intersample peaks in there. So having a metering plugin with that in Sonar, I would use it. It need only be used for a listen through before final render. I guess it's a cpu hog.