Even with Volume Leveling and Headroom Management, clipping can still occur after DSP.
I’m running an FIR-based crossover that stays strictly within 0 dB theoretical gain, yet some tracks still trigger clipping indicators — especially those with high crest factor or intersample peaks.
Interestingly, even oversampling alone can cause post-DSP clipping, which shows that the current fixed headroom model is not sufficient in all cases.
Right now, the only way to avoid it is to manually apply a large static headroom reduction (e.g. -10 dB), but that wastes amplifier gain and dynamic range.
Why this matters
Prevents hidden clipping caused by FIR filters, EQ boosts, or oversampling
Avoids unnecessary fixed gain reduction
Preserves full dynamics and available output level
Especially valuable for multiway and convolution-based DSP setups
Proposed solution
Add an optional Post-DSP Limiter / Auto Headroom Management, based on true RMS or measured signal peaks rather than static DSP gain math.
Possible implementations:
Soft limiter or safety clipper on peaks above -0.3 dBFS
Dynamic headroom adjustment based on measured post-DSP level (true RMS or peak)
Option to apply volume leveling after DSP instead of before
This would make Roon’s DSP chain more robust and transparent — ensuring that no clipping ever occurs after processing, without sacrificing headroom or sound quality.
This is an interesting suggestion for sure. However:
This is a rare problem.
When this matters to the listener, it seems to make sense for that listener to consider their DAC choice quite carefully.
To wit:
Given the above, in context of the many other things people wish from Roon, I am not sure that this will ever get solved within Roon’s solution set. That said, it is, once again, an interesting suggestion.
I understand your point — ideally, recordings shouldn’t exceed 0 dBFS, and yes, that’s rare.
But this isn’t really about bad masters. In practice, recordings vary widely — some peak right at 0 dBFS, others sit 6–10 dB lower. When your playback chain is already near its limits (for example, with low-efficiency speakers or complex DSP), those quieter tracks end up noticeably softer even though there’s still unused headroom available.
Roon’s DSP engine is incredibly powerful — I’ve built linear-phase crossovers for a 4-way speaker — but what’s missing is a way to predict or automatically manage headroom. To stay safe from occasional clipping, I need to reserve about 8–10 dB of headroom for all tracks, which effectively reduces output power by a factor of about six. That’s quite painful with any power amplifier.
A simple peak normalization option wouldn’t “fix bad masters,” it would just help make better use of the available output range, without altering dynamics. It could live alongside the existing loudness-based volume leveling — just a different use case and place in output chain.
I realize this is a very niche feature request, closely related to advanced DSP use cases.
However, it doesn’t seem particularly complex to implement, and it would significantly enhance the value of Roon’s DSP capabilities — which, despite their niche nature, have already been implemented impressively well.
Well, what you propose is impossible to do in real time, since the signal amplitude of the future isn’t known at present, unless you introduce significant latency to be able to ingest and analyze a whole track before starting it, or god forbid, introduce any dynamic limiting algorithm which really causes loss of fidelity when active.
Consider for a moment, that Roon does all it’s DSP with 64 bit floating point precision, gaining several orders of magnitude in headroom between any introduced artifacts and human audibility thresholds, so a few bits of additional headroom can easily be wasted without loosing sleep over audible effects.
Face it, you’ve simply chosen inadequate components for your system needs, and that can’t easily be fixed automagically.
Having 8-10 dB headroom left on your digital chain isn’t going to do anything bad - particularly at 24 bit or more where you have way more dynamic range than is practically needed.
So I would say adjust the gain let the power amps clip if you get to dBFS.
Mostly you won’t get there - and if you do it’s likely to be very loud.
Probably better than extra limiting when you mostly won’t need it.
I actually have a similar setup in my studio. My speakers take AES at 24 bit, we set the speaker gains so dBFS is slightly louder than we would ever want - and then use a digital volume control in the RME interface.
Thank you, your advice is a very practical solution at the system level.
However, I fall into the category of users who can’t simply increase the gain by an extra 10–20 dB (for some recordings), because the 8-channel DAC with integrated preamp stages and the power amplifier input sections are matched for 0 dBFS.
I’m considering a circuit-level modification of the power amplifier input stages, but that would be quite a pain. And I prefer to retain as much of the full D/A conversion scale as possible.