Support and General Use > Audio Playback, Database and Playlists
Enabling 24bit-44.1/48/88.2/96khz playback - iAudio and others?
soap:
--- Quote from: jaylee on September 29, 2010, 09:58:55 AM ---
--- Quote from: soap on September 28, 2010, 08:26:43 AM ---
It IS enabled. Not hardware 48 playback, but 48->44.1 resampled playback. Do you have a specific audible issue with it as it stands now or a theoretical distrust of it?
--- End quote ---
i don't trust resample like 48<->44.1, that's not integral, deteriorates sound quality.
--- End quote ---
I'll repeat my question.
Do you have a specific audible issue with it as it stands now or a theoretical distrust of it?
saratoga:
Regarding other sample sizes:
We already use full 32 bit precision for everything. If theres ever a target with a 24 bit DAC it will work as is with rockbox.
Regarding other sample rates:
We currently resample everything using linear interpolation which results in a lot of distortion/aliasing at higher frequencies. We could improve this either by using better interpolation (cubic spline seems like a good choice, but some kind of IIR based approach might work too) or by supporting more sample rates in playback.
If we wanted to support more sample rates in playback a few things would have to change:
* EQ This will probably work at 48kHz fine, since thats only 9% different. We'd just have to scale the band center frequencies by 9%. It probably won't work at 32 or 96k though, so it would need to be disabled for really wierd sample rates.
* Crossfade This needs to be disabled if either of the tracks has a different sample rate
* DSP Engine All the NATIVE_FREQUENCY macros need to become variables and we need to hunt through the code and make sure theres no hidden assumptions about sample duration.
* Gapless Transitions between different sample rates probably cannot be gapless, so we don't really have to do anything here. They'll be a click when you reclock the DAC anyway
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amishfury:
--- Quote from: torne on September 28, 2010, 08:57:56 AM ---No we aren't. By the Nyquist-Shannon sampling theorem, you only need the sample rate to be double the highest audible frequency. So, unless you can hear above 22KHz, there's no need to use a sample rate higher than 44KHz.
--- End quote ---
there is a very good argument for higher samplerates though... the closer you get to the upper end of the frequency range supported by the samplerate the more aliasing... higher quality DACs of course handle that aliasing better than lesser DACs but also higher samplerates eliminate much of the aliasing simply by having the increased resolution
of course these arguments don't generally apply to portable players where the most common preference is to sacrifice some audio fidelity for smaller file size on top of the fact that most portable players are not exactly designed for samplerates higher than 44.1KHz
torne:
--- Quote from: amishfury on November 02, 2010, 01:05:44 PM ---there is a very good argument for higher samplerates though... the closer you get to the upper end of the frequency range supported by the samplerate the more aliasing... higher quality DACs of course handle that aliasing better than lesser DACs but also higher samplerates eliminate much of the aliasing simply by having the increased resolution
--- End quote ---
See the sampling theorem again: there's no aliasing at all as long as the sample rate is at least double the signal frequency. Aliasing doesn't start until you *exceed* that. Human hearing even in the young stops around 20KHz, so 44.1KHz sampling is sufficient for playback purposes.
soap:
I believe amishfury is attempting to refer to problems caused by the low pass filters on non-oversampling DACs having the tail of their roll-off in the (upper part) of the audible range.
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