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Why Do Cell Phones Suck For Music/ Audio Processing and Playback?

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SPL Tech

Member
Joined
Nov 28, 2006
I wouldn't call myself an audiophile, but I do have a pretty good set of headphones I use while running and I spend the extra time needed to set an EQ to make them sound really good. However, what I've noticed is the iPhone (in my case the 6s) just outright sucks for audio processing--specifically, the applications available that play mp3s on the phone. I am completely unable to find any EQ app (paid or unpaid) that can allow me to increase some of the bands (mostly bass and treble) without clipping and distorting the signal. Even adding +3dB at 64 Hz. causes distortion on some songs. At first I thought the headphones just weren't as good as I thought until I tried using them on my computer. With Windows Media Player I am able to add more than +10dB @ 64 Hz. without causing distortion and clipping the signal.


So, the question is why? Computers and phones dont have actual preamplifiers like a mixer or graphic equalizer used for professional audio would, it's all software. As such, why does no one have a solid app in the app store that can amplify different bands appropriately without clipping the signal and sounding like complete crap? Even apps that claim to be "studio grade" and intended for "processional use" (yea, like pros would use a phone for processing audio....) still completely suck, even compared to WMP which is an outdated, consumer level playback program.
 
Pure speculation, but I have to wonder about the internal format of each platform. The phones, as pretty much closed devices, may be optimised for out of box experience, with no headroom for any tinkering. The problem with digital representation is you hit a hard upper limit. If you boost anything, you run a risk of hitting that with clipping. If this is the case, reduce everything else instead. The possible downsides to this approach are increased quantisation noise and reduced dynamic range. Why WMP might not run into the same problem, I don't know.
 
Could also be the bitrate at which the songs were recorded/streamed, i have a 5S and although the sound is somewhat low (international laws) with the default headphones the quality is very decent if i sync 320kbps or FLAC MP3, zero distortion even if i bass/treble it up. Last time i checked the iTunes store used AAC 192kbps as standard ?
 
Pure speculation, but I have to wonder about the internal format of each platform. The phones, as pretty much closed devices, may be optimised for out of box experience, with no headroom for any tinkering. The problem with digital representation is you hit a hard upper limit. If you boost anything, you run a risk of hitting that with clipping. If this is the case, reduce everything else instead. The possible downsides to this approach are increased quantisation noise and reduced dynamic range. Why WMP might not run into the same problem, I don't know.

Reducing other bands on the EQ does not prevent the one boosted band from clipping. If I add +3 dB. to 64 Hz and -3 dB. to all the other bands, the signal still distorts at the same volume level as if I add +3dB to all bands. I DID find one program that has a dynamic limiter which prevents severe clipping. Basically, when it's enabled, there is a maximum about of gain that can be applied to any one frequency band and adding more beyond the limit does not increase the frequency in that band because the software limits it to prevent distortion. It does a good job preventing the signal from clipping, but I am finding that it does not allow me to apply as much bass and treble as the headphones are capable of producing. In effect, the apps on my phone cannot produce as much bass and treble as the internal amplifier in the headphones are capable of producing (my headphones are wireless Bluetooth), whereas the computer is able to produce enough gain in any one frequency that the headphone's amplifier becomes the limiting factor near maximum volume.

Anyway, the issue you speak of is exactly why preamplifer gain exists. As you boost one frequency you need to reduce the gain on the entire EQ to prevent clipping. Modern [professional] programs do this automatically by reducing the gain as the software detects clipping. However, NONE of the iPhone programs I've found do this with the sole exception of the one app I mentioned. I cant event find one app that has a real VU meter to see how close I am to clipping the signal.... The apps just allow you to adjust a few EQ bands and that's it, but it's ultimately worthless because the second you start messing with the bands you clip the signal and it distorts.

The difference between the phone and computer is absolutely night and day. It's like listening to a $40 set of headphones vs a $350 set. It doesent even compare.
 
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Reducing other bands on the EQ does not prevent the one boosted band from clipping.

The point was not to increase any bands at all. If you want +3 dB in one band, it is equivalent to reducing 3 dB everywhere else, leaving the band of interest at 0 dB. This is kinda manually doing what you mention later about preamplifier gain.

Good software should also use a higher resolution (more bit depth) for intermediate processing to prevent distortion from creeping in.
 
The point was not to increase any bands at all. If you want +3 dB in one band, it is equivalent to reducing 3 dB everywhere else, leaving the band of interest at 0 dB. This is kinda manually doing what you mention later about preamplifier gain.
That's not really how graphic EQs work though. They have a Q factor which determines the width of each band. In professional audio apps the Q factor is adjustable, in simple iPhone apps it's not, at least not in any of the apps I've seen. When you adjust a band it adjusts a certain frequency group. Adjusting the entire EQ up or down does not simply increase the volume up or down, it creates spikes and dips in the bands. In effect, an EQ with +3dB at 62 Hz and a flat response everywhere else is not the same as an EQ with +0dB at 62 Hz and a -3 dB response everywhere else. They will sound completely different. You can see this by setting the EQ flat and then setting all sliders to -10 dB. It will sound completely different because all though the EQ appears flat, it is anything but. It's only flat on the frequencies you adjust, but the interlining frequencies between the bands will not be flat.
 
because its a damn phone, the same reason why it doesnt have an dslr lens hanging off the back of it to take amazing hd pictures, and the same reason it doesnt have 20TB of storage. it is a phone, needs to call / text / check fb for the masses.
 
Hey guys, quick in and out on this one.

Conventional audio devices use DACs to change the digital electronic music into analogical audio that we hear. Most of us are aware of this, but there is also another way to do DAC without discreet components (Resisters, Inductors, Capacitors, etc) and thats by keeping it all digitial.

Digital Signal Processors (DSPs) are used in cell phones and other small electronic devices to perform the digital to audio conversion. Think of it as taking digital signals and just using a bunch of transistors to create the audio wave form. No need to clean the audio when you can switch x10 faster than the audio is being rendered at. So as a result you get a relatively clean sound but no "depth". Furthermore, the faster the sampling rate to produce the signal, the more information you can output. Faster switching and measuring of the audio signal creates a cleaner output. Also since its all transistors, you can't get the same dB as you would with discrete components. Higher power load transistors do not switch fast, but slower transistors can switch more power. So you have to play in happy medium for all with this design choice.

This is the extent of my knowledge on the subject, but to me I find DAC systems (that even a consumer motherboard can provide) will have better sound than a really good DSP.
 
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That's not really how graphic EQs work though. They have a Q factor which determines the width of each band.

I have to admit I'm not familiar with what method actually gets used in practice. The filter you're describing is what I know as the old school type, as more easily implemented in analog days. In the era of software and digital filters, there is no need to do that, unless you want to replicate it. Filters can much more easily match an "ideal" response.
 
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