Saturday, 25 January 2014

MUSINGS: The Audio PC / Music Server / HTPC (Basics, My PC, and Some Generalization)

I'm in the process of finishing up some measurements - I think many will find it interesting in the days ahead. I'm going to let that article percolate a little first however. This week I thought I'd spend some time discussing/considering the computer system for media consumption; a bit on both the hardware and software aspects, and hopefully putting together bits and pieces I've done over the past year in the process. As you can imagine, there's quite a lot to cover...

In the hopes of a reasonable summary, here are the 3 main component functions of computing devices (almost everything is a 'computer' these days) in the media room:

1. Media Client or "Media Player" or "Transport (to a DAC)": The computer acts as a "front end" to feed your DAC/receiver/amp and or TV/monitor. Of course you can also have a good internal sound card (like the ASUS Essence STX), and I presume few are using the motherboard's sound output (usually of comparatively poor quality). You interact with the computer with whatever player software to control which tracks/files are played, how to "fast forward", "stop", "pause", etc. in much the same way as the disk spinners. Specialized "computers" may have hardware controls for this like a play button on the front panel of the chassis, maybe a specific remote control unit rather than using a generic keyboard. On-screen display on the monitor/TV can be as simple as a web-based interface like the Squeezebox server or the customary GUI of something like iTunes, foobar, JRiver, etc.

This transport/playback task can also be delegated to streaming devices of course (which are essentially little computers inside). For example, the Squeezebox familyMeridian Sooloos, Cambridge Audio streamers, Naim NAC-N, Linn DS, Bryston BDP-2, LUMIN etc... That's the higher end, but on the low end, you have things like the WDTV Live previously measured. These devices usually need to be connected to some kind of server for one's music (although some devices like the Squeezebox Touch can act as its own miniserver) and many can access Internet based streaming media like the hundreds of Internet radio stations around the world, Spotify, MOG, Pandora, SiriusXM, maybe Beats Music in the days ahead.

Although audio streaming may be all one needs, for those who "want it all", the pinnacle of the media room computer is the HTPC (Home Theater PC) with both audio and video playback capability. The HDMI interface has become the de facto digital audio-video cable to do it all. Multichannel 7.1 hi-resolution audio is essentially universal these days with modern HDMI interfaces, and 24/192 sample rates can be sent to a decent modern AV receiver with no problem. Some AV receivers will also accept DSD. IMO, multichannel PCM is preferable because DSP manipulation of the audio stream is an essential part of getting multichannel right... Good bass management (some SACD players are able to do this in DSD), channel reassignments (eg. 5.1 fold down to 4.1 system), room corrections, are all easily done in the PCM domain and would require a DSD-to-PCM conversion step if you're sending out a DSD bitstream. This limitation of DSD is a big one in multichannel and unlikely to be solved any time soon... If ever...

As for video with the HTPC, it's trivial to achieve 1080P. 4K resolution can be reached with the current HDMI 1.4 specification. As of early 2014, I suspect the market feels little compulsion to buy the current generation of 4K TV's and one would achieve little benefit apart from early adopter bragging rights (and spending quite some money in the process!). First and most painfully obvious, there's no content nor even a clearly announced means of media distribution (it looks like 4K/UHD Blu-Ray is still in development). Second, I'd suggest waiting for the wide availability of HDMI 2.0 which would allow 60Hz 4K frame rates (I see that Sony has released firmware for certain TV models already, DisplayPort can already achieve 4K/60) since there does at least seem to be some push towards >24fps movies and if I'm getting a "next-gen" TV, I'd want that. At this time, the real benefit I see from 4K is finally being able to watch 3D in full 1080P on a passive display. I'm waiting for one of those to hit an affordable price range in the 80+" size & HDMI 2.0 :-).

2. Media Server: This is the "back end" where you store your music (videos, movies, pictures). In this day and age with easy connectivity, there's nothing to keep the server in the media/listening room. Many people have opted for NAS storage and since there's a CPU inside the NAS unit, it could also run server software like Logitech Media Server, or the scads of UPnP/DLNA servers. In fact, in a home where there's wired ethernet throughout the house, you can easily have the server computer or NAS on a different floor/room. In my experience, a gigabit network can transfer data just as fast as many inexpensive high-capacity hard drives (50-100MB/s is normal with gigabit ethernet using standard Cat5e cable). A great benefit to this is that you can keep the playback machine (computer or streamer) simple, low power, cool and silent without having a bunch of hard drives running in the same enclosure while listening to your music or watching movies.

3. Mobile Control Apps: Although not specifically the computer itself, the ability to use one's smartphone or tablet computer has been a great boon to the usability of digital media playback. No longer do we need to turn on the TV to select albums, or select video. These days, I still use iPeng and SqueezePad on my iPad to control my Squeezeboxes (Logitech Media Server). Squeeze Commander works fabulously on the Android devices. Cover artwork adds to that overall presentation.

However you want to mix-and-match the functions above, there are a myriad of options based on what OS you choose, which server software, and how the media is being played. The hardware itself can be any combination of devices like NAS, laptop, desktop, network streamers, etc. It is this fantastic flexibility that can be a source of frustration to those starting to enter the computer audio world. Commercial companies are obviously interested in capturing part of the market with devices such as the Aurender computer systems, Sooloos Music Server System (see Streaming products). Not surprisingly, these turn-key products are usually Linux based, low power, relatively slow (often Intel Atom CPU, sometimes ARM based), and generally quite a bit more expensive than something one can put together with standard commodity parts. The greatest thing about true technological innovation in the marketplace is the deflationary price pressure - take advantage of it if you can! Have a look at Computer Audiophile for some ideas on building one yourself.

As a "case study", I figure it might be of interest to show the system I'm currently running...

I basically have an all-in-one box that's a server to my Squeezebox devices all over the house, a digital audio transport to my TEAC UD-501 DAC, as well as full HTPC functionality to the ONKYO receiver and 55" LG TV for movies and videos.
Hmmm... Maybe should clean up a few of those cables in there. Logitech Unifying receiver sticking up front for the keyboard.
HTPC quietly doing it's thing in the corner... The smaller box beside the computer is a CyberPower CP1500PFC UPS. Keyboard is the Logitech TK820 with touchpad.

Main hardware components:
- Fractal Define R3 midtower quiet case (2 quiet case fans - rear exhaust and front to cool HDs)
- Seasonic X-400FLII fanless 400W PSU
- ASUS F2A85-V Pro motherboard (HDMI 1.4)
- AMD A10-5800K APU (integrated AMD Radeon HD7660D GPU)
- CoolerMaster Hyper 212 Plus CPU cooler
- 16GB (2 x 8GB) Kingston Hyper X Blu DDR3 RAM
- SiliconImage Sil3132 port-multiplier eSATA card
- 128GB OCZ Vertex 3 SSD boot drive
- 2 x 3TB WD Red for music library
- 1 x 2TB WD Red for video/movie library
- 1 x 1TB WD Green for web server data
- 1 x 1TB WD Green for misc data backups
- LG BH12LS35 Blu-Ray reader/writer - playing the occasional Blu-Ray & audio ripping

OS: Windows 8 Pro / 64-bit

The AMD A10 CPU has worked well for me in the past year. In fact, I undervolt and underclock it slightly to 3.5GHz to keep it cool and quiet with the CoolerMaster CPU heatsink/fan. I haven't measured, but the CPU power consumption would be substantially less than 100W.

Notice that the HTPC isn't built with necessarily the newest generation hardware components. In fact, much of this was put together more than a year ago. Unless some "disruptive" killer app were to be released that needs much more computing power, I suspect this would be all I need for the next few years (clearly the push to upgrade is slowing). Audio processing doesn't take much power. At times I will turn on SoX minimal phase upsampling à la Meridian for the Transporter (see VirusKiller's thread), other times play with JRiver's PCM to DSD conversion à la EMM Labs, or try out a convolution room correction filter with foobar - never has the AMD A10 felt underpowered for these tasks. With a better audio room since moving into the new house in late November 2013, I've been quite aware of the slight noise the computer makes. Recently, I replaced the power supply with a fanless model (the Seasonic 400W) which lowered the noise a bit.

At this point, the computer is still slightly audible on account of the spinning hard drives (softer than the Panasonic Blu-Ray player in use). The problem with upgrading the room significantly (my ambient SPL is <30dB(A) at night) is that you also have to get the other parts up to spec as well ;-). I guess I can look at moving the hard drives used for video and web serving over to another computer on the home network to drop a few more dB's.

I have 6TB of space for all the music (all backed up on another machine over the gigabit network). The library consists of stereo PCM CD's (~3000+ albums) and hi-res downloads and rips (~250 albums). I have some multichannel 5.1 music (~100 albums) in PCM format taken off my DVD-A/SACD/DVD/Blu-Ray/DTS-CDs. All the PCM music encoded losslessly as FLAC.

Finally I have a small ~30 album collection of DSD stereo music which I know are either sourced from genuine DSD recordings or analogue transfers (ahem... no Norah Jones Come Away With Me type faux DSD, thanks). These DSD recordings are stored as .dff because I like lossless DST (Direct Stream Transfer) compression. As I noted months ago in my SACD/DSD Musings, I do not like the fact that one currently cannot have both tagging and compression. Without compression, DSD files are unnecessarily large, wasteful, and ultimately inelegant IMO. DST compression brings them down smaller than the size of average 24/96 files and I make sure that the filenames I choose can be easily parsed for album, track number, artist, and title. I don't have many DSD albums on the server so it hasn't been difficult. Despite the ongoing hoopla around DSD, I remain sceptical that DSD will have much traction unless a simple foundational issue like a fully featured, modern, file format is addressed. (Actually, I just suspect there's not going to be maintained traction simply because DSD doesn't bring much to the table...)

I'm not going to say much about the video playback since much of it is just family videos with some Blu-Ray rips I made for demo purposes when people come by to visit. The AMD A10 APU has a built-in graphics processor and the HDMI out of the motherboard works well to send multichannel 24-bit audio and 1080P video to the AV receiver.

Player Software:
I see audiophiles can really get heated about this. For those who have been around this blog for awhile, you'll know that I did not measure any difference between bit-perfect software players whether with Windows or Mac OS X using the TEAC DAC. This was the same for PCM and DSD. Furthermore, the "audiophile" player JPlay made no difference and in my opinion risked audio errors with extreme settings like unnecessarily low buffer space. Since I do not claim to have particularly "golden" ears (I'm in my 40's now), I likewise hear no difference between these players.

On my HTPC, I currently have 3 audio "servers"/players running:

1. Logitech Media Server (aka Squeezebox Server back in the day). All my stereo PCM music is streamed out to the Squeezebox units I have scattered around the house. In the listening room, I have the Transporter on the rack. BTW I do have a few 24/192 albums in my collection, but for most DVD-A rips where I have the physical copy, I usually downsample to 24/96 anyways. There are many "fake" 24/192's out there (just like 44kHz upsampled DSD) and even for those that are genuine hi-res recordings or analogue transfers, there's rarely content related to the music itself above ~40kHz (not that anyone would be able to hear it!), so I figure there's no point wasting space.

Because I have 16GB of RAM, I've set up a 4GB RAM disk (free one for personal use) and modified my configuration for LMS to put the database there. Really speeds up library searches - almost instantaneous.
Logitech Media Server (LMS): Some Billy Joel or Benny Mardones anyone? :-)

As mentioned above, I use SqueezePad, iPeng, and Squeeze Commander to control the server.

2. foobar2000: Fantastic, flexible, free software I keep running 24/7. The library keeps track of all my multichannel PCM music and the default output device is to the ONKYO TX-NR1009 multichannel receiver via HDMI using WASAPI. FoobarCon Pro is my preferred controller software off my Nexus 5 or 7; cool that it also has panels to view artist bios and track lyrics so you can sing along as the surround sound plays :-).

Foobar playing 24/96 5.1 Crowded House DVD-A rip.

3. JRiver Media Center 19: Again, I keep this running 24/7. Works beautifully and fully featured with a huge number of customizations and options. Although it has many advanced features like the ability to realtime upsample PCM to DSD64/128, I'm actually not using this for any of my PCM playback... JRiver is dedicated to feeding the TEAC UD-501 all my DSD music in native ASIO! As I mentioned above, .dff files cannot be tagged natively but JRiver can keep an internal database and has the ability to parse the filenames to "recreate" the album data, sort artists, tracks, etc. Furthermore, it decodes DST without any problems even when in the past I found issue with foobar's DSD decoding plugin. Nice :-). I don't think there's another software package that will do all this in such a hassle-free fashion with a lovely presentation at a very reasonable price (~$50USD).
A peek at the DSD library in JRiver using filenames and path to reconstruct tagging information.

JRiver provides the free Gizmo app on the Android for playback control. It works well but functionality is more basic than something like the Squeezebox controllers above. However, the really cool part is that JRiver can transcode and play music and videos to said Android device via Gizmo. A reminder that JRiver isn't just for music but works well as a full-featured "Media Center" for all your A/V needs.


So that's a glimpse into how I'm currently using my HTPC. Perhaps some of this could be useful for your setup as well.

Now about audiophilia and computer audio...

To close off this blog entry, let's talk about the computer in an audiophile setup; specifically achieving excellent sound quality. I know many people believe that all kinds of arcane software tweaks such as turning off unused processes like printer services, BIOS tweaks, etc. are necessary to ensure good sound (something like this). Much of the OS tweaks probably do no harm and some of these recommendations may have been useful at one time (like a decade ago); I just don't think much of this is relevant any more or makes any difference. As far as I can tell, jitter under high CPU load is not an issue even with a simple TosLink off a motherboard as I showed here so I hope nobody falls for the "it causes jitter to be worse" explanation unless demonstrable at the level of the DAC output. Others in the past speak of using low power CPU's for audio (I haven't seen as many proponents these days). For me, the good thing about a more powerful machine is a speedier user interface, faster file scanning for the server, and also the opportunity to use DSP like convolution room correction filters without the machine breaking a sweat. Of course, you'd want a more powerful computer for video playback. Some others even advise against using lossless compression. Seriously, does anyone still actually believe a processor unintensive task like FLAC lossless decoding will cause enough electrical noise/interference to make it sound worse than a WAV file especially played off an external DAC!? (I certainly hope ideas like this will become just as bizarre as the belief in greening the edges of CD's 20 years ago.) Sadly, over the years, various audiophile magazines have promulgated much speculation and disinformation without checking facts or consulting with common sense (much less science/engineering).

Let's keep it simple - IMO, the main ingredients of a good computer audio setup:

1. Keep the computer sonically quiet! As few fans as possible if not fanless. Laptops are great for this - something like a MacBook Air or Ultrabook would be fantastic for example given how quiet they run. If you can, relocate noisy hard drive servers to another room with wired network (I consider wireless too unreliable for my taste and can be strained by high-resolution data rates).

2. Keep the computer away from your audio gear to reduce EMI/RF from entering the analogue path.

3. Get a good DAC. External units are great because they can be placed with your other components and isolate the computer as in point 2. Make sure you're using the best driver especially with PCs such as bit-perfect ASIO instead of going through the Windows Mixer to ensure bit-perfect output. Also, jitter has more to do with the quality of DAC than anything you fool around with on the computer side. From my measurements posted around here over the last year (eg. look at the TEAC UD-501 PCM results), a good modern asynchronous USB (or ethernet streaming) is generally better than SPDIF (coaxial/TosLink) due to lower jitter (J-Test results better but for the most part I doubt it's audible).

4. Things to not sweat about: cables - just make sure your power cords and interface cables work and look good enough to you in your room. IMO expensive cables may look good and convey a sense of authority, but please do not equate aesthetic value (eg. jewellery) with function (ie. "better" sound). Specific make/model of computer - again, this is aesthetic and so long as it runs your choice of OS, you're good. (No, I do not consider Apple computers as somehow better sounding.) OS - Mac, Windows, Linux, whatever so long as your server/player software runs well on it. As I showed here, different laptops and OS's connected to the same asynchronous USB DAC results in exactly the same analogue audio output. While I can't vouch for every computer, so long as bit-perfect output to the digital interface is assured, there's no need to fret. Player software - Again, see my bit-perfect measurement posts here, here, and here. Find one that has all the features you need and achieves bit-perfect output.

If you have the above down pat, then by all means tweak to your heart's content! Just don't break anything...

I just realized I've been building my computer audio library since 2004 (10 years already!). For those new to computer audio, I suspect all of this could sound overwhelming (and I'm sure I missed some important points). Stick with it, play around with it - it won't take long to pick up. No matter what I do with the computer setup, without doubt, the most time consuming bit of all has been to make sure all the music is tagged properly and named in a consistent fashion (try Mp3Tag). Keeping the directories clean, using the same filename for cover images, and ensuring bit-perfect rips (try dBPowerAmp CD Ripper) do take time and effort; this is as expected since it's all about the music, right? Despite all the effort, high-resolution digital audio is as good as it gets for the audiophile who values high fidelity and the convenience in accessing all your music with a few search keystrokes is undeniable.

It's a great hobby with many avenues to explore. Just don't forget to listen, and enjoy the music :-).

PS: Backup regularly.

Friday, 17 January 2014

DEMO & MEASUREMENTS: What does a bad USB (or other digital audio) cable sound like?

Okay, so the other day I was installing my new BenQ BL2710PT monitor (reasonable monitor for a decent price) and as I was rummaging through my old cables, came across a very old USB 6' cable that I probably got free when I purchased an old Samsung laser printer back in 2001 during the transition from USB 1.1 to early USB 2.0.

This cable is the thinnest, most flexible, likely most poorly shielded USB cable I have; in other words, about as "bad" as it gets when connected to a quality USB DAC which expects to operate in high speed USB 2.0 mode without completely failing... Behold the "Bad Cable":

Plugging this cable into my desktop ASUS Essence One provided the opportunity to demonstrate just what a poor USB cable does to the sound... I'm sure this is "old hat" to those who have experience with digital audio, but for those who haven't, have a listen...

I recorded 1 minute of a freely available track from Jason Shaw called "Pioneers" from here off the Essence One fed into my EMU 0404USB to the usual Win8 laptop using a good quality cable versus the flimsy one above.

Good USB 2.0 cable - well shielded 12', ferrite core on both ends of this specific cable:

"Bad" USB cable as pictured above - poorly shielded against interference and incapable of transmitting at bit-perfect high-speed data rate to the ASUS Essence One:

Even though SoundCloud recompresses the uploaded FLAC audio, I'm sure you can appreciate the obvious errors in the "Bad Cable" sample. (You can press play back & forth between the two samples to A-B them if you want.)

What you're hearing is what happens with digital error (ie. not bit-perfect), similar to watching digital TV with the occasional data error leading to macroblocking and bad pixellation as in this sample found off Google (notice the blue stripe due to digital error):

It's worth noting a few characteristics of this poor cable as it pertains to sound:

1. Poor digital cables leading to digital errors sound like brief pops or occasional static (assuming they do not completely malfunction). They're similar to the errors you get when ripping a CD without something like EAC or equivalent. Sometimes, you'll hear very brief dropouts. Depending on the data packet disrupted, occasionally they will occur in only one channel but not possible for this to happen consistently in a single channel. Remember that although asynchronous DACs have the capability to buffer, hence improve timing and lower jitter, they do not (at least not in the case of the Essence One with the CM6631 USB interface as far as I can tell) necessarily error correct or ask for a packet resend. The more data error, the less the amount of "normal sounding" music will be heard. Obviously if the data error occurs every few minutes, it might be difficult to detect, but if it happens frequently, it's not subtle.

2. A poor digital cable does not result in overall level changes in the song... This is not like analogue distortion that can consistently alter the volume level or change the dynamic range uniformly or periodically.

3. Similar to the above point, poor digital cables are not capable of changing the overall tone of the sound. There is no such thing as a digital cable capable of acting as a "tone control", making certain sounds "brighter" or "warmer". A passive digital cable is not capable of acting with some kind of frequency filtering mechanism.

4. Poor digital cables do not consistently do anything to the soundstage. A poor digital cable cannot make a voice or instrument sound "distant" or move it "forward", or pan the soundstage to the left or right as a whole or in relation to other components of the music.

5. Bad cables cannot cause speeding up or slowing down of the data transfer. Poor digital cables therefore cannot cause sporadic or consistent timing issues like warble (speed up/slow down pitch changes), "pacing", or rhythm problems.

6. The concept of cable "break in" makes no sense with digital audio cables. If it carries data accurately when plugged in then the only problem that can happen in time is corrosion at contact points or reactions such as oxidation of the metal over time. This can only lead to transmission errors as demonstrated above, not some magical improvement due to "break-in".

7. I was reminded here the other day about the measurements with a poor RCA cable I used as coaxial SPDIF last year. Indeed, if you use a very poor, unshielded RCA cable paying no attention to the expected 75-ohm impedance specification with an SPDIF digital interface that's not galvanically isolated (eg. coaxial SPDIF of the ASUS Essence One in that case), noise can be introduced into the system. However it does not take extravagantly priced cables to make things right (an inexpensive 6' <$20 decent shielded cable from a reputable company will do). As always, noise can be introduced into the analogue domain with any electrical connection (or just being careless like putting your DAC right on top of a noisy desktop computer), so it's not really an issue with the digital system itself.

You might be curious how the 2 USB cables measure in terms of jitter...

Surprised? As you can see - not much difference at all! If you monitor the realtime FFT for the J-Test, you will see errors "popping up" with the bad cable due to bad data transfer, but in between, the jitter plots are essentially indistinguishable! This is expected... For an asynchronous DAC like the ASUS Essence One, jitter rejection is handled very well by design and there's nothing the passive cable can do about that.

Now I'm sure there will be a number of folks who disagree and hear various effects in the list above (see here, here and here for some interesting perceptual accounts and/or creative writing). The thing is, where is there decent evidence to show that passive digital cables (and I'm talking here not just of USB but also the SPDIF variants like coaxial or TosLink) sound different if they're error free (a.k.a. bit-perfect) and built to specifications (assuming no issue with analogue noise as in item 7 above)? I've never seen manufacturers come up with anything of substance... Or hobbyists/DIY guys show/demonstrate verifiable claims... As usual, I'm happy to change my mind if some kind of objective evidence exists since I personally have not subjectively heard a problem except as demonstrated above with digital errors.

(Digital cable summary from a number of months back for those who might have missed it. Recent post on EETimes blog on this.)

Recommended album:
- Have a listen to Babatunde Olatunji's posthumous 2005 album Circle Of Drums. This is one of the best Chesky albums I have heard. The drums sound fantastic with wonderful tonality and sense of 'space' around the instruments; a lovely exploration of African drums and rhythm. Unless you believe you can hear the difference between 16-bit noise floor and that provided by SACD, IMO there's no need to buy the SACD because it appears to be a 44kHz PCM upsample (here's the Master List). There is a multichannel mix on the SACD which sounds OK but derived from post-processing. An impressive sounding and quite enjoyable record for those interested in world music nonetheless!

Relax and enjoy the music!

Addendum (January 20, 2014):
For the sake of completeness in answering Frans' comment below, here is the J-Test result with the TEAC UD-501 using the poor USB cable vs. good one:

The TEAC uses quite a different asynchronous USB implementation than the Essence One. It appears to utilize a TMS320 DSP processor rather than "off-the-shelf" USB interface like the CM6631(A). I'm wondering if this interface is actually more robust since the digital errors were even less obviously audible.

In any case, using a different DAC, the jitter test remains unchanged; two examples now of how an obviously poor USB cable does not appear to affect the jitter from asynchronous DACs in terms of the analogue output (which IMO is the only important measure since that's what we hear!).

Sunday, 12 January 2014

MEASUREMENTS: Google Nexus 5 and Nexus 7 (2013) audio quality...

Mobile space is where it's at these days for major consumer computing innovations... New advancements in wireless communication, low power, more speed. Amazing just how fast progress is being made! It was only in early 2000 that the 1GHz barrier was broken for consumer desktop CPU's, 2005 was the first mainstream dual-core x86 CPU (both AMD X2 and Pentium Extreme 840). Now, we've got >1GHz quad-cores with 1080P screens on handheld devices in less than 10 years... (Yes, I know an ARM CPU isn't directly comparable to the x86, but still impressive nonetheless!)

For fun, I wanted to run the current Google Nexus' headphone output through the measurement gear to have a look at what some mobile folks are listening to...

I. The 'Victims'

A. Google ASUS Nexus 7 (2013 2nd generation model) - Android 4.4.2 "Kitkat", stock ROM, 32GB storage


I'm really liking the smaller 7" form factor for tablets. This tablet acts as a fantastic controller for my Squeezebox Server using Squeeze Commander (my stereo PCM library streamed to Squeezeboxes), Gizmo for JRiver (all my DSD music to TEAC UD-501), and FoobarCon Pro for foobar2000 (multichannel PCM to ONKYO HDMI receiver).

It feels fast with a 1.51 GHz Qualcomm Snapdragon S4 Pro. Have had it for a few months, but I've actually never listened to the headphone output until tonight. Sounds fine with my Audio-Technica ATH-M50 in terms of bass weight and resolution. It can't drive the M50 loud so you'd want to use higher sensitivity headphones.

B. Google / LG Nexus 5 - Android 4.4.2 "Kitkat", stock ROM, 32GB storage

Nexus 5 lying on the Panasonic QE-TM101. Love the QI wireless charging!

The nice UPS man dropped this off on Christmas Eve. I gotta say, compared to most of what we see in audiophile land, this is true value! Plenty of features, one of the fastest current mobile CPU's, great 1080P screen as well squeezed into 5". This phone has replaced my Samsung Galaxy S2. Some people have complained of poor battery life; I find that it's fine with Google Now turned off.

Like the Nexus 7 above, I didn't have a listen with headphones until tonight. Maximum volume seemed a bit higher than the Nexus 7 using the ATH-M50 which was a surprise. Again, in sounds pretty good. "Dead Already" off the American Beauty soundtrack maintained its usual bass impact through the headphones. Demanding loud tracks like "To Victory" off the 300 soundtrack wasn't as defined but not bad.

II. Results

Okay, pretty straight forward setup here... Android accepts FLAC without issue, all that was needed was to copy over the calibration and test files via USB and off to the test bench. Of course, all DSP/EQ/bass boost off. I left the phone and HSPA+ data on for the Nexus 5 and WiFi on for Nexus 7 as the most likely situations in daily use.

Nexus device headphone out --> phono-to-RCA cable --> EMU 0404USB --> shielded USB --> Win8 laptop

Unfortunately I don't see any specs on the output impedance for either of these units so it's hard to discuss headphone matching.

RightMark summary:

Totally unfair, but I threw in a couple of 24/96 Squeezebox measurements on there - the SB Touch which represents a good upper level consumer device, and the Transporter which is in the audiophile league of audio performance.




III. Summary

Well, there's not much surprise here. The built-in DAC's on these devices are limited in fidelity. The DAC circuitry is integrated into the tightly packed SoC which includes the CPU and GPU, situated in close proximity to wireless communication hardware for WiFi, BlueTooth, G3/HSDPA/LTE transmission...

Having said this, I was actually a little surprised to find that the smaller Nexus 5 phone performed a little better than the Nexus 7! The frequency response was more even and there was less harmonic distortion found... Evidently, it's not the size that matters.

Neither unit could benefit from 96kHz sampling rate - looks like it's downsampled to 44kHz.

Although neither device could deliver beyond about 16-bit dynamic range, it was interesting to find that 24-bit data resulted in slightly lower noise floor (you can see this easily with the J-Test graphs). Interestingly, the jitter modulation pattern was visible with both devices for the 24-bit J-Test suggestion high jitter levels (but really, who cares?).

Obviously, both the Squeezebox devices measure and would sound superior to these portable units.

I've found it curious the recent developments in smaller audiophile gear. Those small USB DACs for example seem to preform reasonably well - the AudioQuest Firefly, Meridian Explorer both look good and measure well for those with small desktop space or want a high quality DAC on trips. It's quite clear the success of the Light Harmonic Geek speaks to the market in such devices.

What I'm not so "bullish" about are those "audiophile iPods"... Stuff like the HiFiMan units, or the Iriver Astell&Kern (oh yeah... I almost forgot about the Pono - supposed to come out in early 2014 so says Neil Young - I'll believe when I see it!). I find it hard to consider portable players as anything other than convenience items. They're meant for headphone use of course, but even if they performed excellently (it looks like the Astell&Kern AK100 measures well), what's the point? The more resolving headphones that can benefit like the Sennheiser HD800 are open design with poor noise isolation and I cannot imagine many folks crazy enough to walk the streets or take the subway with them on. To make matters worse, the high end headphones usually require more power and that's going to really drain battery life assuming the "audiophile iPod" even has the oomph to provide enough volume in the first place.

Bottom line... For the purpose of the commute, these measurements of the Nexus 5 and 7 appear totally fine assuming you have a good pair of efficient headphones. From my collection, volume was adequate with the Apple white earbuds, JVC HA-FX40 IEM, Sony MDR-XB500, Koss PortaPro, but not really good enough for the Sony MDR-V6 or aforementioned ATH-M50 in a busy environment... Don't even bother trying open headphones like the Sennheiser HD800 or AKG Q701. However, surprisingly the Nexus 5 drove the Audio-Technica ATH-AD700 better than I thought.

Addendum: (January 29, 2016)
A number of years since measuring the Nexus 7! However, I started doing some impedance measurements... Using 20-ohm and verified with 300-ohm loads, 1kHz sine wave, I'm seeing an output impedance reading of 33-ohms. Not good but much lower than that of 100-ohms reported here!

Room Measurements First Steps (Stereo): Paradigm's PBK-1 and Room EQ Wizard...

PBK-1 mic about to record some subwoofer beeps and boops...

Alas, the Christmas & New Years holiday season is behind us. Each year, I'm amazed at just how quick everything goes by... Never enough time to enjoy the music among the rest of life's other demands.

I didn't get much time off but I did manage to try out some room measurements in the evenings when the kids were in bed.

For all the time I spend thinking about audio components, it's unfortunate that I've spent too little on the room itself. With the new house and room, I hope to change this. Remember folks, as much as "we" (audiophiles) might obsess over the quality of our DACs or (some) might sweat over things like (supposed) cable differences, without question, the quality of the transducer/room interactions trump essentially any of the other factors in the audio system given the quality of electronics these days. Frequency variations in the audio room can fluctuate wildly, on the order of magnitudes of difference compared to the rather trivial differences between DACs for example. It might not be cool or sexy to speak about bass traps, diffusers and absorbers versus DSD or fancy cables made of Unobtanium, but this is what's ultimately going to make huge differences.

Of course, room measurements are not a new topic... Many excellent web pages have been devoted to this. For room treatments, check out Ethan Winer's RealTraps site. Also, Mitchco's excellent room calibration article not long ago documented his procedure nicely.

I'll have the years ahead to add hardware around the room, but as a first step, I'm just going to try the simplest of "room corrections"; EQ'ing frequency response using DSPs.

I. Paradigm PBK-1 (Perfect Bass Kit)

Paradigm, Anthem, and Martin-Logan are owned by the same parent company and share similar technologies in terms of room correction. The Paradigm Signature and Martin-Logan subwoofers have built-in DSP units that can be programmed using a PC with the PBK kits; based off of the Anthem Room Correction algorithms. As the name implies, this kit is used for bass correction only as it applies to the subwoofer. Audio levels still should be checked with the rest of the system in order to make sure the sub integrates well.

As of this writing, the current version of the PBK software (2.01) is not compatible with Windows 8 (even with compatibility settings)! I had to dual boot the old laptop used for measurements with Windows 7 and run the software through that... Paradigm, please update the program!

The software was easy to run and essentially self-explanatory. A USB cable connects the included PBK microphone (shown above) to the PC, then another USB connects the PC to the subwoofer. Each microphone has been calibrated and identified by serial number in the software. The program is capable of measuring multiple locations (max. 10) in the room to smooth out the bass response. Since I'm most concerned about the "sweet spot" and want to limit the potential of suboptimal calibration, I took 5 readings all around the central seat and the 2 adjacent seats. The program then will run thorough the calibration algorithm and show a screen that looks something like this...

You have the option to adjust the DSP crossover frequency point which in the graph above I've set to 160Hz (default is 250Hz) as well as how steep the filter should be. This gives you some customization options (not much).

The red curve was what I got in the room. As you can see, I have quite a dip at around 65Hz (red) which was correctable to some extent (purple). The peaks (eg. around 30Hz correlated to the calculated lateral mode for the room size at 29.1Hz using this online calculator) were easier to correct, and it's nice to see good frequency response down to 20Hz with this sub. (Check this link out for the room mode math calculations.)

The program will automatically upload the new settings to the subwoofer and away you go... Very simple calibration to do.

II. Room EQ Wizard (a.k.a. REW)

REW (5.01beta) can be downloaded free off the Home Theater Shack website - just need to register. It's just an amazing piece of Java code for the audio enthusiast.

About 3 years ago, I purchased a calibrated Behringer ECM8000 mic (I see they don't sell these calibrated any more at this site). This microphone has served me well over the years and put to good use here again (note that I actually measured it a little lower at ear level sitting on the couch than the picture below):

Using the EMU 0404USB as measuring ADC, here's the raw room response with just the PBK settings in place for the subwoofer integrated with the main front speakers (1/6 octave smoothing) using the TEAC UD-501 DAC:

Although I don't know if I fully trust the Behringer mic below 30Hz and above 15kHz, it's good to see frequency response down to 15Hz. Again we see the room mode around 29Hz. The deep blue line represents the eventual target curve we're aiming for based on the default REW house curve (for those looking for the excitement of a bass-induced thrill ride, try this target curve). For those looking for more based on home theater wisdom, check out this link on House Curves and more!

Letting REW perform its own EQ from 20-200Hz plus a few small adjustments on my end resulted in this mathematical prediction of room response:
Much more controlled on the low end using 7 parametric EQ settings (you can see the numbers above) plus 2 settings at 3kHz and ~11kHz to roll off the top. Total of 9 EQ settings were programmed into the Behringer DEQ2496 applying the adjustments in the digital domain and looped back to the Transporter for DAC duties.

Since some frequency boosting is involved, I reduced the DEQ2496 digital output levels by 6dB and double checked with some really LOUD music to make sure the EQ settings did not lead to clipping. The 1997 Iggy Pop insane remaster of The Stooges' track "Your Pretty Face Is Going To Hell" off Raw Power is a good one - average dynamic range value of 1dB for the album! If the DSP processing doesn't clip with that track, it's probably not an issue with >99% of my music.

Since it's always good to confirm that the EQs are actually doing what they're supposed to, here's the actual measured room response with & without the Behringer DEQ2496 played through the Transporter as DAC (measured at a higher level on a separate day):
Before REW EQ:

After REW EQ:

OK. Nice real-world confirmation that the EQs are doing what they're supposed to. The predicted results checks out even with a different DAC (measured with TEAC, confirmed with Behringer & Transporter - I knew from previous measurements that the Transporter is very close in frequency response to the TEAC)!

Looking Ahead...

Like I said above, I have many days ahead to make this room "work" better acoustically. EQ'ing so far is just the "quick and dirty" first step at this point focused essentially just on volume equalization (although I guess the PBK might be doing more in the algorithm). I haven't even begun to try using the Audyssey MultEQ XT for multichannel yet (in the Onkyo receiver), nor room treatments...

Regarding room treatments, there's much to do! Here's the measured waterfall spectral decay plot in REW:
15Hz - 20kHz
With EQs in place, the decay time of course remains high:
15Hz - 20kHz

Plotted at 500ms duration down to 40dB with 1/6 octave smoothing, I'd really love to see more uniform steeper decay (<<300ms). Bass traps in the corner and absorption panels to the sides at the first reflection points could do the trick. Hmmm, maybe this will be a project for Spring Break - whip out the saw, stapler gun, make some wood frames, grab fabric and a stash of Roxul Safe'n'Sound :-).

There is also the issue with time alignment as addressed by mitchco using Acourate and convolution filters. That's another level of tuning I'll have to leave for another day! So much to do, so little time...

For now, the subjective sound quality has improved. There's already notably better control to the bass notes from Rebecca Pidgeon's "Spanish Harlem" (to use a well known audiophile favourite). Time to just sit back, relax, and enjoy some tunes! I've got a couple of ideas for tests coming up.

Musical selections recently:
- Daft Punk's TRON: Legacy soundtrack (2010) on Blu-Ray was stunning! I missed the movie in the theaters and rented the 3D Blu-Ray the other day... The movie itself was OK but the surround effects and techo score really made the movie an audio feast.

- The Eagles. Love 'em or hate 'em, I reacquainted myself with the multichannel DTS version of Hell Freezes Over (1997) the other night. IMO another fantastic multichannel release from the earlier days of surround sound when DTS was releasing their DTS-CD's (this was also my first concert DVD). That live ambiance really shines through as if you're sitting in the audience that night. The guitar work and percussion sound great in the new system on "Hotel California" - especially the bass impact. I saw them in concert about 3 years back and that too was a blast.

- Speaking of bass... I don't often buy modern pop recordings but I did enjoy listening to the recent album by Lorde - Pure Heroine. Fantastic job by the 16-year-old from New Zealand. The current Top-40 'hit' "Royals" gives a nice taste of the cavernous bass found throughout the album ("400 Lux" is another to check out). Have a listen to this album through a system with clean bass down to 20Hz and see whether you think you need a subwoofer :-).

In the "Answer To What Question?" and "What Were They Thinking Of?" files... I was looking at the recent CES2014 announcements in the usual audiophile watering holes and found this:
Esoteric Grandioso D1 Monoblock DAC
Given the level of performance of even modest DACs these days, I really can't imagine what would be the reason to go monoblock with a DAC. Seems like doing this could make things worse (channel desynchronization? need for DAC matching?) and at a significant expense ($22,000 each, not to mention all those extra cables!). Small price to pay to feel grandioso I suppose. As usual, would love to see the measurements for a pair...