Dave brat headphone amplifier
We should have that experience together. I wanted the guys to feel like they were giving something to the project beyond just playing. In other words, the Missouri-bred, Denver-based frontman wanted to make the band disappear along with him—out in the middle of the desert at first, and then deep in the woods. The result is the aptly titled Tearing at the Seams , a vivacious and inventive full-band record, with significant contributions from all eight members of The Night Sweats.
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The board was designed using Kicad, etched by himself and put in a machined enclosure. All the production files can be downloaded from his website so you may produce it within a day. That xiph study is flawed. Your comment would be more convincing if you actually said what you think they missed, instead of just repetitively stating that the Xiph articles are wrong.
The Xiph article is based partly on those studies and partly on well-proven signal-processing math. Put simply, when you sample a signal that is close to but below the Nyquist limit of your sampling frequency, you get a beat frequency in your sample as a result of the interference between the sampled signal and the sampling rate.
The beat amplitude diminishes the further away from the limit you go, and becomes virtually unnoticeable at around 0. Much more than that would be pointless, though. The USB box featured here is for playback, not recording. To put it simply, if you downsample ITB to Think about it. That means you can actually fit two 20khz tone cycles in the same space of one 10khz tone at one of 2 phases: Synced and half a cycle behind, meaning stereo sound above 10khz can only be completely out of phase, or completely in phase, no in between, and anything more accurate cannot be expressed due to the limitations of the sample rate.
This is still a problem down to around 4k in terms of hearing a difference with a trained ear. THIS is why higher sample rates are better for playback. Not because of higher frequency sound, but because of much more accurate imaging. For the record, Gen0 is completely wrong. Sampled audio above 10 kHz can be completely out of phase, or completely in phase, or anything in between. Phase is not quantized. Higher depth and frequency for sampling is useful due to alignment, phase, noise floor, and overshoot concerns.
Cost of chips from for example Farnell is close to 40e. Then add passives, connectors and enclosure but I would say way below e. I like the project, the build and the fact he shared all the files. Having said that, I wonder about the perceptible difference in sound quality. I switched to an off the shelf DAC Fiio E17 because I wanted spdif and USB and the hiss is gone, though you can introduce it again by increasing the gain on the headphone amp stage, that says it all really. The optical output is digital, so unless something goes seriously wrong with the hardware or software it will transmit the correct sequence of bits.
Into a DAC box like this, powered from batteries. Should be economical enough to run off batteries, or use rechargeables. Then the electrical noise only starts from the output of the DAC. SPDIF does get rid of hiss, but from my experience it has its own problems. Supposedly there are some differences because both ends of a SPDIF link have to have accurate timing hardware and not all do. It is silly to compare headphone jacks a powered speaker output to a digital link digital sound only. You will feed that digital audio signal to a digital to analog, which is, guess… an amplifier!
The internal soundcard will round off the bit level, split up the sampling, incorporate noise, etc… it will sound smeared and muffled. An external soundcard, otoh, will sound crisp, clean, and clear with snappy dynamics and high contrast.
Your output device speakers, headphones, etc… will also matter, but only in proportion to how clean the rest of the signal chain is. Sound is anything but simple and YMMV!
Seriously though, just pluging the cheap 4 dollar USB sound card it into my externally power USB hub instead of my computer helped my audio quality. On the other hand, mainboard audio outputs are top of my list of sucky, crappy sound outputs, so improving these is easy and likely. Sound quality is the least of just about any motherboard manufacturers worries when they have GHz range signals to tend to.
But compared to a normal motherboard sound chip I now which I prefer. Thanks, HAD! Might be time to have another look at a multi input multi output mixer I guess. Some projects are never intended to be finished.
Sure the design has room for improvement but then you need to wonder just how much improvement will make an audible difference. Cheap DAC chips often can be a serious bottleneck, especially the all in one jobs on a typical motherboard.
Not at all! The problem which makes them suck is the design around the chip, not the chip itself — exactly like this DIY design.
If anything, he should have understood that the design around the chip is key. Improve the design until the chips become the bottleneck, THEN use high end chips to improve it if necessary. Using high end chips on a poor design gives poor results regardless it merely increases the price of the BOM, nothing more. Motherboards are a prime example of that.
I have most of the top level dacs — that record producers use to when they master records. Not because they can produce perfect square wave, but because they have the best sound. The schematic has too many wires crossing, could use some labels. Nice build, but it seems more like a non-hifi, non-high-end approach.
Can the buffered outputs drive headphones? But besides that it should work? Would be interesting to see what kind of results one could get. Yep, it is not easy on a single-sided board. But I have no facility for producing a double-sided board. There are however R resistors on the outputs here. Ok great, thanks! Commercially, yes. Companies such as Apogee first one that comes to mind, though there are a number of companies have multichannel hi-res DACS.
The design considerations mentioned by [AnarKIT] and [heh] would really need to be dealt with before doing any scaling though… first things first. There are plenty of integrated 8 channel audio DACs. My on board sound is really bad. There is a high pitched squeal over the output. It is so loud that even with music set to max volume it can still be heard. There is also some noise, but this is masked by a music signal.
This seems to have a major flaw, from datasheets the PCM samples at 48 khz, 16 bit. I dont know if it passes through i2s at a higher sampling rate but it seems like a waste to then run it through a more expensive PCM for Digital analog conversion this chip is made for up to khz 24 bit DAC so why not just use the PCM as the DAC it has built in digital analog conversion.
It uses a 5v headphone amp IC and a 5v linear reg decoupled on the input and output with ceramic and electros. Of course it sounded horrible with all the associated electrical noise of hard drive and fan motors injected into the sound.
I fixed it by inserting a 60 ohm resistor in series with the 12v supply, and sinking the supply current though the audio jack from the motherboard.
You can use all the decoupling you want, but the fact of the matter is that the onboard sound has its own analog ground plane with supply rails generated from a linear reg. Since this is single ended audio, ripple in the ground potential has just as much as an effect as ripple on the signal.
The result is that when using 32 ohm headphones it amplifies the background hiss of the motherboard DAC to a point where it is audible. I have in turn fitted a potentiometer to the output is that is left to attenuate the gain produced by the amplifier most of the time. I can turn it back up again when I want to jam out. I have built this circuit twice now with two different layouts, and am having the same problem. Using the OPA I am finding that with identical negative feedback resistors R in this case, on both circuits I am getting a 0.
I am inputting a sine wave through audacity tone generator Hz 0. Because of this I have had to adjust the gain resistor to enable the output voltages to match. The issue is the same on my second circuit, but different layout. Has anyone else experienced this? Is this simply a resistor tolerance problem which has happened by coincidence twice? Or a op-amp chip issue? Or other? Any ideas? Is there any possibility to get this DAC to run at khz?
Maybe change the PCM to something else? Maybe there is a different chip for the I2S conversation available which has the same footprint as the PCM…? Could someone give me a link s where I can learn more about DACs? Could you help me, please? One being Software Defined Radios …..
I am looking at a similar project but I also need a microphone input to the codec. In doing research I contacted TI and learned something I was not expecting. They have acknowledged that I2S standard is not limited to 16bit for a 12MHz clock but they claim that 16bit is still all that this part will do.

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If ever a drummer summed up a band's sound, it's Tre Cool. His bull in a china shop approach perfectly suited Green Day' s attitude-drenched early days, while his hard and heavy yet restrained when needed latter day style is matched by the more mature sound found on the band's more recent output. This week the pop punk brat upstarts turned stadium rock heroes kick off a trio of enormo UK dates, reinforcing their position as 21st century rock superstars. In recognition of their undoubted standing, we're looking back at Tre's finest moments behind the kit.
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