Even Better PC Audio v. empty your wallet

My goal with the FIIO is to have better sound quality than that which comes out of the sound card in my PC. That sound card is built into the motherboard, so I don't expect much from it. For the <$200 USD I spent for both devices, it's within the range I wanted to spend on something like this and also offers a convenience of portability with the E7. I don't think I'm at the point right now to even consider $600-$700 for a headphone amp. :-) I appreciate the recommendation though.
 
I have an observation and a question about the playback of FLAC audio through a software player (like winamp or foobar 2000) but before I do that, I want to explain how the audio device is connected in case this matters.

Dekstop PC > USB cable > E7 (DAC docked in E9) > E9 (Amp) > 1/4" headphone out.

I first began playback of of a bit-accurate FLAC rip of my CD. I first fired up winamp to playback the FLAC file (it was my default). The audio is shown as going through windows mixer as a "USB Audio DAC" device and not my normal integrated sound card (which is to be expected). I noticed the occasional click/pop during playback which I have not noticed before. I felt like something was wrong with either the E7 or E9.

I switched to foobar 2000 and switched the playback output device to the WASAPI : Speakers (USB Audio DAC) device driver. I set it to 16-bit and so far the playback sounds better. Is this just a placebo or is it possible that foobar 2000 is better at FLAC playback when compared to winamp? I've not heard any pops/clicks and I feel like the music is clearer...even clearer when compared to when compared to the output DS device driver playback option. I notice a slight volume decrease when using the DS playback option when compared to the WASAPI so it could be the volume change which is tricking me into thinking it sounds better.

The other thing I noticed with the WASAPI playback is that it only plays back the audio from foobar2000 and none of the windows sounds play (which I like).

So far at the comfortable volume levels I like to listen at, I hear zero hiss in the amp. Even if I pause the playback and turn the volume all the way up, I hear no hiss. I can detect a slight sound as the volume knob is rotated, but it stops as soon as I'm done rotating the volume. Is there a general opinion on what to set the digital gain at on foobar2000 when playing back audio like this or does it not matter?
 
WASAPI should bypass the Windows volume controls completely. It also locks the audio device exclusively so nothing else can use it. You should have the volume in Foobar up all the way and use the volume knob on your headphone amp.
 
Thanks Stereodude. Is the WASAPI the ideal playback option for quality? You recommended this to me a while back when I was trying something else.
 
Stereodude said:
Oh please... If they decode correctly they'll all sound the same.
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You're right of course, I'm sorry to have upset you. It's just like with speakers and head phones. All speakers sound the same because they only need to move the speaker membrane back and forth a few times/second to reproduce the right frequencies. If it's done correctly they all produce the same sound.
 
fb said:
You're right of course, I'm sorry to have upset you. It's just like with speakers and head phones. All speakers sound the same because they only need to move the speaker membrane back and forth a few times/second to reproduce the right frequencies. If it's done correctly they all produce the same sound.
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Apples & oranges... :rotfl:
 
fb said:
Not really, both the apples and the oranges introduces distortions into the sound chain. The only difference is that one of them is more obvious.
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Yes really...

There's no such thing as a perfect transducer (speaker driver) which is one of the reasons why speakers sound different from each other. There are perfect decompressor (software algorithms). We can easily prove that Foobar's FLAC handling is perfect by converting a FLAC to a .wav and comparing the audio data of the original wave to the decoded wave. If the output from the decompressor is perfect there's no way another software program could sound "better" unless your definition of better happens to include non-perfect operation where you happen to prefer the artifacts of its non-perfect operation.
 
Has it been proven anywhere that foobar2000 can perfectly decompress a FLAC file?

What is your software preference of audio playback on your laptop when using your DAC/amp? Do you use foobar2000 or something else?
 
Handruin said:
Has it been proven anywhere that foobar2000 can perfectly decompress a FLAC file?
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I'm sure somewhere, but why take someone else's word for it when you can prove it for yourself?

I just tried it. I converted a .flac file to a .wav with the official windows FLAC software and then with Foobar2k. I used RIFFStrip to toss the headers and then did a fc /b on the two raw PCM files and SHOCKER!!! no differences encountered. :rofl:
What is your software preference of audio playback on your laptop when using your DAC/amp? Do you use foobar2000 or something else?
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I use pretty much use Foobar exclusively.
 
Thanks for doing the comparison. I want to try a few of those with winamp when I get home later. I realize that's a full file decompression comparison, but is that how the audio is sampled while streaming/playing in a software-based player? Does the entire decompression occur before playing, or does it happen during playing?

I know this is a noob question, but when I'm using a USB sound card (like the E7), what format is the music transferred over the USB cable to the sound card for the DAC? Is it send as PCM?

Is this the approximate understanding of the path:
FLAC > foobar2000 > PCM > E7 DAC > amplification > analog headphones/speakers
 
Handruin said:
Thanks for doing the comparison. I want to try a few of those with winamp when I get home later. I realize that's a full file decompression comparison, but is that how the audio is sampled while streaming/playing in a software-based player? Does the entire decompression occur before playing, or does it happen during playing?
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It happens while playing at a close to real time rate to keep the output buffer full. The decompressor will have the same output regardless of the speed at which it's decompressing.
I know this is a noob question, but when I'm using a USB sound card (like the E7), what format is the music transferred over the USB cable to the sound card for the DAC? Is it send as PCM?

Is this the approximate understanding of the path:
FLAC > foobar2000 > PCM > E7 DAC > amplification > analog headphones/speakers
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Yeah, that's a reasonable approximation of the signal /format path.
 
Stereodude said:
I converted a .flac file to a .wav with the official windows FLAC software and then with Foobar2k. I used RIFFStrip to toss the headers and then did a fc /b on the two raw PCM files and SHOCKER!!! no differences encountered. :rofl
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But wasn't the CD's played bit-perfect back in 1983? And what was the sound like? I'm glad that there are some open minded engineers out there who knows that there's always room for improvements - or we would still be listening to bit perfect crap. ;)
 
fb said:
But wasn't the CD's played bit-perfect back in 1983? And what was the sound like? I'm glad that there are some open minded engineers out there who knows that there's always room for improvements - or we would still be listening to bit perfect crap. ;)
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I never said that 16bit 44.1kHz audio was "perfect" or couldn't be improved upon.

You however made the absurd and preposterous claim that other software players would sound better than Foobar despite Foobar's perfect reconstruction of the original PCM audio being sent to the DAC. Now after the light of truth has been shined on your audiophile nonsense you switch topics and attempt to turn the conversation to the old, "Is CD quality good enough?" spew.
 
No, but a CD player from lets say 2008 is better than a CD player from 1983. Why? It's the same 1s and 0s coming out of the CD-player. ;)

And I remember one time when an upgrade of Winamp resulted in a radical drop in the CPU usage, so of course different media players have different algorithms that's optimized over time.
 
From some of my readings in the past, I've been made aware that the CD format is still viable and very good even today with regards to dynamic range. The problem is with the recording companies compressing the music to death (loudness war) to make their recordings as load as everyone else because consumers would equate a quieter CD with an inferior product/recording when the opposite seems more true.
 
fb said:
No, but a CD player from lets say 2008 is better than a CD player from 1983. Why? It's the same 1s and 0s coming out of the CD-player. ;)
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:skepo: Well, do you have ABX testing to show that it actually sounds better?

If we hypothetically assume that a CD player from 2008 sounds better than one from 1983 it's probably due to improvements in DAC design. However, this is a total non sequitur to our original discussion where Doug is using the same DAC and you're trying to claim the PC program feeding the DAC makes a difference.

So, to fix your flawed analogy it would require saying that a CD player from 2008 feeding a DAC via SPDIF sounded better than a CD player from 1983 feeding the same DAC via SPDIF. Which would promptly cause me to ask for verifiable proof of those extraordinary claims.
 
fb said:
No, but a CD player from lets say 2008 is better than a CD player from 1983. Why? It's the same 1s and 0s coming out of the CD-player. ;)

And I remember one time when an upgrade of Winamp resulted in a radical drop in the CPU usage, so of course different media players have different algorithms that's optimized over time.
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I would argue that it's not the CD format (redbook audio) but rather advances in DACs have improved over the years. Now with things like EAC, I don't believe the traditional DAC like that of the one in a set top CD player comes into play. Here you're extracting the bit-perfect 1's and 0's into another format (FLAC). Then the FLAC decoding sends the data digitally to a DAC down stream.

Your example of winamp and the improvements in CPU usage have more to do with the codec (assuming) MP3 and not a lossless format. I could be way off, but I am making a guess that the coding and decoding of MP3 is more challenging than that of a lossless format like FLAC because there is a lot of creative manipulation of the audio signal since it's a lossy format. In other words, what is the best way to make this audio file sound good while throwing away and compressing data...
 
Stereodude said:
However, this is a total non sequitur to our original discussion where Doug is using the same DAC and you're trying to claim the PC program feeding the DAC makes a difference.
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I just said that a different program might be better. I can't try it myself since I don't have a USB-DAC. I came with the suggestion since many people claim those programs I mentioned result in better sound. ;)
I'm sure all programs are feeding bit perfect numbers into the USB-port.

So, to fix your flawed analogy it would require saying that a CD player from 2008 feeding a DAC via SPDIF sounded better than a CD player from 1983 feeding the same DAC via SPDIF. Which would promptly cause me to ask for verifiable proof of those extraordinary claims.
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Everything has become better since 1983... Just look at a circuit board from now and then.
 
Anji by Chicken Shack sounds just awesome in the UE 10s, but more modern music with more of a low-end fails a bit. Is this a weakness of headphones in general? Of IEMs?
 
ddrueding said:
Anji by Chicken Shack sounds just awesome in the UE 10s, but more modern music with more of a low-end fails a bit. Is this a weakness of headphones in general? Of IEMs?
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It's hard to say with modern music if it's the headphones is simply the recording itself. There's always the argument for no replacement like displacement, but I don't know how well that holds up for headphones. I do find the bass in my new Grados to be very clean and able to hold up well with the low frequencies. A lot of which has to do with the actual housing. As soon as I put my hands over the open back area, the bass is completely diminished. It's incredible how much of a difference the enclosure makes. Not only is it the enclosure, it's also the ear cups. My point being, with so little room inside an IEM, maybe it's just a matter of physics that makes it difficult to reproduce the lows correctly.

What source are you using to drive your UE 10s and is the music format a lossy or lossless? Is it possible to find a sample CD that plays some test tones in the lower regions to see how well you can hear them as well as how the headphones can reproduce them? What I mean is, if you can't hear the 20-35 Hz range...maybe something is missing in the chain?

If you like the acoustics of Anji, check out a Pandora radio station of Nickle Creek. I know the quality won't be as good as a CD or even a higher quality MP3, but you'll get a good sense of some spacious-sounding and detailed guitars. I enjoy the music by Nickle Creek in general and I've been getting some good hits in Pandora for alternatives to the same genre.
 
Thanks for the recommendations, they are pretty neat.

Found massive error on my part #1: I left the airline adapter (aka loudness filter) on, and that killed the sound altogether.

#2 was I just had it too quiet. I couldn't hear anything outside, and the mids and highs were perfect, but I had to turn it up a couple notches for the low end to show up.

Rodrigo y Gabriela - Stairway to Heaven from their Live in Japan album is truly awesome. That will be one of my listening tests from now on; the detail and energy in the acoustic guitar is fascinating.
 
So I finally got around to rewiring for balanced audio.

Here are my adapters and the 4 pin XLR end.

hpcablerewire.jpg
 
ddrueding said:
That does look great. What wire are you using?
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I used the wire/cable that came with the HiFiMan HE-500's. It's a 4 conductor silver cable. The 1/8" stereo jack came molded on the end of the headphone cable. So, I cut that off about a foot from the end and then put a female 4 pin Neutrik XLR plug on that. I cut off another 12" off the cable and made a 2nd adapter cable with a 1/4" Neutrik TRS jack and female 4 pin Neutrik XLR plug. Last, I put a male 4 pin Neutrik XLR on what was left of headphone cable.

Now I can use the headphones on gear with 4 pin balanced, 1/4 TRS jack, or 1/8 stereo jacks without the need for any 1/4" to 1/8" adapters. I just need to put the appropriate balanced to single ended adapter cable on the end.
 
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