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mitchco

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Everything posted by mitchco

  1. Here is one, maybe too expensive? https://avahifi.com/products/abx-switch-comparator Maybe you just want a switch box? Parts Express may have a simpler one...
  2. @jaaptina sounds good! Do you have a screen shot to share?
  3. Hello, not to take a away from Chris's nice review of the Lumin X1, but not sure what the issue is here. Chris's response is not flat at the listening position. It follows the EBU 3276 standard of flat to 2 kHz and then slopes down at a rate of 1 dB per octave in the high frequencies. We initially tried the Harman target of 20 Hz to about -10 dB at 20 kHz which was a bit too dull sounding given that Chris's space is large, well damped, and the speakers have wide directivity. This included trying a partial correction to 500 Hz and letting the speakers handle above that. As a side note I have written about the Harman curve for both speakers and headphones in a number of articles on this site. We also tried the ITU, B&K and Bob Katz target, all of which is covered off in the range of operational room response curves. This is part of the process of trying various industry standard room response curves and determining which one is the most preferred given ones speakers and room. As you can see in the testimonials, there are a variety of different high frequency roll offs to accommodate size of room, how damp versus live, how far away from the speakers, etc., all play into what is subjectively heard. This is the balancing act between direct and reflected sound in ones room. More lively rooms tend to favour the Harman target (like mine, which I use), whereas well damped rooms in larger spaces trend towards the EBU target with increased direct sound as there are fewer reflections. both of which arrive at the same neutral tonal response at the listening position. This is why there are a variety of target response, Put the target curves together on overlay like I have done on my site, shows all responses roll off in the high frequencies to varying degrees, but all in tight grouping. A/B switching between all 4 target curves is fun and educational as they sound more alike than different. It boils down to a matter of personal preference given ones room and speakers. To be clear, we are not eq'ing the steady state response to be flat at the listening position. None of the targets are flat and we using frequency dependent windowing to differentiate between room sound included below 500 Hz and direct sound above 500 Hz. See DSP Technical Application Note on my site under DSP tab for an explanation of the approach. Kind regards, Mitch
  4. Hi @PorkChop Yes, this will work on a 2.1 system and yes to taming the bass portion of the total signal.
  5. Good points @omid Was there a reason for longer than 10s sweeps? Just curious. Yes, you want to follow the minimum phase response of the roll-off of your speakers. One can also use partial correction if the top end is to one's liking. I would also recommend a small cycle HF window to prevent over correction if using full frequency correction. Typically 6 cycles for the low end and 1 cycle for the top end works pretty well in most rooms without the top end sounding compressed or strident. One could start with 6 dB of correction, gen a filter and then try 12 dB of correction, gen a filter and A/B to see if there is a preference for one or another. For sure, there is filter insertion loss. But one can a) normalize the gain (in JRiver anyway) which brings some of the gain back. b) add digital gain is an option, just watch out for clipping. Interesting about the sub polarity. I have the exact same case in my other article using Audiolense to integrate subs, but I did not invert the polarity and the step response came out perfect. Curious if you have a before and after step response with the polarity switch? How does your system sound?
  6. Hey @omid sorry I missed this. Yes, something not right there... The correction should start at identical times. Mine does. In Audiolense, did you measure with clock drift correction enabled and use separate play and recording streams? Or already sorted?
  7. @kohmeloThanks for your comment. Yes, the reason is DRC changes the amplitude/frequency response in the room and as a result will change the level of room resonances. May want to check your audio path. In my case, I felt the soundstage depth got deeper as there are less room resonances to mask the depth of field in the recording.
  8. Hi @Graham Luke are you referring to TruePlay? "Trueplay then applies a combination of equalizer and filtering techniques to correct these frequencies so your music sounds the way the artist intended it to." This is similar to eq room correction, and very cool! But I see no mention of removing room resonances (i.e. shortening the rooms decay time)...
  9. Hi @JR_Audio Juergen, thanks! I hope you are well. I did not try that experiment, but next time I am measuring, I will. It does work dynamically with a Sense control in the digital domain, but is based on knowing your rooms impulse response ahead of time. So the software already knows the rooms frequency response and decay time before it processes the music... if you catch my drift 🙂 My hypothesis is that the frequency response will measure the same at different SPL's. Anecdotally, that's what my ears tell me while listening with Room Shaper on listening at different SPL's. Kind regards, Mitch
  10. @sdolezalekSee this article on three acoustical issues that digital room correction (DRC) can't correct: 1) Speaker Boundary Interference Response (SBIR). Agreed. Speakers like the Kii THREE and D&D 8c with their cardioid designs help deal with this issue. However, my research and results show that some DRC software packages, (that don't boost eq), do a decent job in dealing with this. 2) Strong Early Reflections. Agreed, DRC does not address this acoustic issue. 3) Long Decay Times. Agreed. DRC will not address long decay times, for example, in a bare room. This requires passive acoustic treatments. However, this is where Room Shaper comes into play as it targets and removes room resonances that have long decay times at low frequencies (i.e. think below 100 Hz). Thierry's Room Shaper software also takes the "boxiness" sound out of one's room, which is another type of room resonance, also reducing its long decay time. DRC does not do this. Hope that helps explain.
  11. Thanks. I linked two articles that explain room modes, standing waves, and room resonances, but a GIF of a standing wave may be helpful, agreed. I have listened (and reviewed) the Kii THREE and D&D 8c in my room before. Their cardioid design does a great job of minimizing Speaker Boundary Interference Response (SBIR). However, below the room’s transition/Schroeder frequency, the room is in control as per Dr. Floyd Toole’s (and others) research that I linked to in the article. In other words, regardless of speaker, room resonances, especially below 100 Hz, are determined by the physical size, shape, and construction of the listening room. As mentioned in the article, the worst resonances are the ones with long decay times, which is what Room Shaper addresses.
  12. Home Audio Fidelity - Room Shaper Review Home Audio Fidelity’s “Room Shaper” is an innovative digital signal processing (DSP) software product designed to even out low frequency room modes in your listening environment. Low frequency room modes that have uneven decay times can have a negative impact on one’s listening experience. This can be heard when certain bass notes (i.e. frequencies) take longer to decay than other bass notes. Often called room resonances. In this review, I show how to setup and configure Room Shaper, objectively compare the before and after decay times and provide a subjective listening experience using music to hear what was removed from the room. I must say upfront that I am impressed with this innovative DSP software product that lives up to its claim of “as if the room disappeared.” Preamble Some technical info to help folks understand what problem this product is trying to solve. It involves some physics and how a room can impact the sound quality, especially the low frequency response of loudspeakers in a room. If you prefer, one can skip to the objective measurements or subjective listening sections. Typical sized living rooms and home theaters are dominated by modal resonances in the low frequency region. Unless one has acoustically designed and constructed a room using proper room ratio’s to distribute the room modes evenly, virtually every listening room has uneven bass response. From the linked article: The key phrase is, “it is the excessively long sound decays of the modes that make them stand out and more audible.” It is akin to the resonance one hears when blowing air across a Coke bottle for example, except at a much lower frequency. Technically, one can use Digital Room Correction (DRC) products to equalize the minimum phase response at the listening position. Further, the DRC products I have reviewed provide time domain correction by time aligning speaker drivers and applying excess phase correction. Room Shaper’s algorithm is different than DRC as it targets low frequency decay times that are longer than others. By using time domain correction, shortening the long decay frequencies to be the same as adjacent frequencies results in an even decay time over frequency in the low end. Much like changing the shape of ones room to a preferred room ratio. Room Shaper also assists in taking out the “boxy” sound of a room up to 600 Hz. What does “boxy” sound like? You will be able to hear it on the recording I made later in the review. It should be noted that Room Shaper does not produce a digital FIR filter like DRC products where filtering is a linear transformation applied continuously to the whole signal. Room Shaper is based on events detection and modifications of only some portions of the input signal. Room Shaper can complement any DRC product as it is targeting a different issue (i.e. reducing low frequency decay time) than DRC addresses (i.e. frequency and time domain correction). In my case, I use Audiolense and Acourate. Of course, one can simply use Room Shaper as a standalone VST plugin as DRC is not a requirement. I am using JRiver MC25 as the host for the plugin. To further illustrate the point, our listening rooms have a transition frequency where room resonances dominate the frequency response and the speakers are no longer in control. Here is a graph similar to the one linked above but showing the transition or Schroeder frequency: From Floyd Toole’s Audio Science article. It is calculated based on one’s room dimensions where room resonances take control of the low frequency response regardless of the speakers being used. I like this Room Mode calculator as you can move the cursor along the frequency scale and it will output a tone at that frequency. If your computer is hooked up to your speakers (careful with the volume!), you can hear the resonances in your room by hovering the cursor over the modes in the graph. It is an ear opening experience and great for training ones ears to know what to listen for. Good digital room correction systems can even out the low frequency response at multiple listening positions below a room’s transition frequency. Unless one has an acoustically designed room, with good room ratios, DRC is almost mandatory as one can’t (easily) change the physical dimensions of the room. What about bass traps? While bass traps do absorb low frequencies the issue is below Schroeder frequency one needs to literally stuff the room with bass traps in order to have any significant absorbent impact below 100 Hz. The downside is, aside from the dollars spent and the number of traps in one’s room, bass traps are not “surgical” in nature and also absorb frequencies above the affected region (i.e. above the rooms transition frequency) to the point of having a near anechoic chamber for a range of frequencies. Usually the upper bass and lower mids are too absorbed. Having worked in some really absorbent rooms like studio control rooms and critical listening environments in the past, too much absorption really sucks the life out of the music. There are a few industry guidelines that specify how lively a listening room should be based on room size for listening conditions for the assessment of sound program material and methods for the subjective assessment of small impairments in audio systems. OK enough tech talk. Hopefully this illustrates the problem well enough to understand how Room Shaper solves a particular acoustic problem that the majority of us suffer from. From Floyd Toole’s research, bass subjectively accounts for 30% of how we judge speakers sound quality. Setup, Measurement and Configuration After downloading Room Shaper, the plugin can be installed in any software program that supports VST plugins. I am going to install it in JRiver Media Center 25. While one can place the VST anywhere on the hard drive, I copied Room Shaper to C:\Program Files\Steinberg\VstPlugins. In JRiver, click on Tools and select Options. In Settings, select DSP and output format. At the bottom left, click on Manage Plugin-ins. Navigate to the location where Room Shaper VST was copied to. Here are a few screen shots from JRiver to help with the procedure: Once installed, it should look like this: Room Shaper should be moved up as far as possible in the DSP ordered list. One can simply drag Room Shaper to move it up the list so that after the default JRiver processing, Room Shaper is next to process. We also want Room Shaper to process the full range signal, so click on Options to the right of the plugin and select, “Process independently of internal volume:” Since I have been using Room Shaper for a while, we can see mine is already configured. Let’s reconfigure so folks can see the steps of what needs to be done. A calibrated measurement microphone, like the UMIK-1 is required as we are going to take left and right speaker measurements using the acoustic measurement software package REW with the measurement mic at the listening position, at ear height, where we would typically listen. There are several guides for taking acoustic measurements. In this article, I wrote a simple section on “set up to take measurements” (scroll down a bit) is a quick primer. REW also has a few guides, including using UMIK-1 for folks that are new to REW. From Home Audio Fidelity’s manual: This procedure describes how to perform room impulse measurements by using REW and an external player to play the measurement (sweep) signal. By using this procedure, it is possible to include a digital correction while performing the room measurement: Download the sweep signals Add them to your favorite player Activate or not digital correction in your player (convolution for instance) Start REW and configure your microphone Select "Measure" option and configure the panel with following options SPL start Freq: 20 Hz end freq: 20 000 Hz length: 256k "use acoustic timing reference" "wait for timing reference" the other parameters can be left as default Press "Start measuring": REW will wait for a sync signal that is embedded in the downloaded sweep signal. As REW will play its own test signal when you press "Start measuring", be sure REW’s own test signal is not audible. The easiest way is to choose the internal sound card of your computer as an output device (via REW Preferences menu) and put your computer volume at zero. Play the left channel sweep signal (_L_refL) from your player and check that the level measured by REW is in the acceptable range (make sure to stop the player after the file plays). Rename the measurement (Left for instance) and export it as a WAV file. See screen shot below. Repeat the procedure for the right channel by playing the right sweep signal ( _R_refR) Rename the measurement (Right for instance) and export it as a WAV file After you have measured both channels, the REW screen should look similar to mine: The two measurements are now ready to be exported. To Export the impulse responses: The following dialog appears: Use the same settings you see above. Export both left and right channels to a project folder you can create on your hard drive at whatever location to contain the impulse responses. In my case, since I loaded Room Shaper VST into JRiver, this is where I will configure the plugin by importing the exported room impulses by clicking on “IR Left” button on Room Shaper. A file dialog will open and select your left impulse response: IR Left is now loaded. Import IR Right. Then click on Configure: The processing may take a minute or so. Once completed, we are ready to listen to some music. But first, let’s take some objective measurements so we can understand the impact of Room Shaper. Objective Measurements This is a unique DSP software product and I was unsure of what would be the best way to objectively measure the impact. I did use REW’s waterfall graph and it indeed shows a measureable difference in reducing certain frequencies, but does not really relate the musical impact of what the processing is doing in the time domain. Another approach is comparing music waveforms over time. Audibly, it allows one to hear the impact using music. It is simple procedure to perform as one can record the digital output of a music player, once with Room Shaper in the signal path and another recording without. Here I am using JRiver to play music and using Audacity to record the digital output: I recorded some samples from Madonna’s Ray of Light album, “The Power of Good-Bye” as an example since it has sustained bass notes that allows one to hear what is being removed from the room. I recorded about 90 seconds and is used on the basis of fair use for the purpose of evaluation and research. I am not doing this for financial gain. Please delete these music samples once you are done with the evaluation. Note my Lynx Hilo uses a multi-client ASIO driver, which allows me to use multiple client applications so one app can play music and the other recording the output. The next step is to load the two recorded stereo tracks into one’s favorite audio editor to perform some basic digital audio editing. The idea is to line up each stereo track so that they are in perfect sync. Then invert one stereo track and mix down all tracks to a summed stereo track. Whatever is left over, is what is being removed by Room Shaper. This procedure is often referred to as differencing. One can review the detailed procedure in this article if you are interested. Here is the difference file so you can hear what is being removed in my room. The level has been normalized so you don’t have to turn it up super loud to hear. Of course, in the room, it is at a lower level relative to the direct sound as can be seen later with the dB scale spectrograms: Madonna delta Room Shaper (15 MB wav) Madonna delta Room Shaper (13 MB zip) Here is the frequency spectrum of the difference file: Only frequencies below 600 Hz are being processed (i.e. the ones with the long decay times) and is reduced in level at the output. One can see the low frequency peaks that are being removed, along with the “boxy” room peaks between 300 to 500 Hz. It helps to listen over headphones (i.e. to remove your room) and ideally one has headphones that go deep enough in bass response to hear what is being removed from the room. If you listen closely you can hear consistent bass resonances being removed and on certain bass notes quite a bit of resonance is being removed, especially the longer the bass note sustains, the more is removed. Another aspect one can hear is the “boxy” sound of the room being removed. Remember, what you are listening to are room resonances and no direct sound. It may take a moment or two to tune into what one is hearing. To help visualize this, here are a couple of charts that illustrate the acoustic issue and what exactly is being reduced in level. The top chart shows the spectrogram (i.e. decay time) of my room (in milliseconds) over frequency and the dB level is in color. The bottom chart is the corresponding frequency response measured at the listening position. Left speaker: Right speaker: Even with a smooth frequency response at the listening position, one can see the long decay times in my room below 100 Hz in the spectrograms. If you look closely, you can see there is longer decay times in the 300 to 500 Hz range as well. That frequency range is what gives a room a “boxy” sound quality to it. It is the combination of room ratios and how lively the room is, which in my case is pretty lively as there is little absorption in my room. Other than a couple of bass traps, a half dozen broadband absorbers, quiet curtains, and a throw carpet (with double underlay) between the speakers and leather couch, everything else is drywall, glass and hardwood floor which dominates my room. So what does Room Shaper do to the long decay times below 100 Hz? Here is the left channel spectrogram where on top is the room and the bottom is with Room Shaper on. I have put a red square around the area to focus on when you compare the top to bottom: The red block encompasses frequencies from 10 Hz to 1 kHz and time frame from 250 to 500ms. Note in the top diagram that resonance at around 70 Hz is a solid greenish blue color (i.e. about -20 dB in level) and does not diminish in level, well past 500ms. In the bottom graph we can see that has been reduced to be in line with adjacent frequencies plus all the way down to 10 Hz has been significantly reduced (i.e. -20 dB of reduction). In the top room spectrogram, you can also see several solid darker blue lines from over 100 Hz to about 300 Hz stretching from 250 to past 500ms. Again, in the bottom pic using Room Shaper, we can see that area has been reduced in level to be at similar levels to adjacent frequencies that are not long decay times. The right channel is more of the same: If you look at 10 Hz in the top spectrogram and out to just before 400ms, one can see a pretty high level resonance (about -15 dB in level at 400ms with a twinge of green). In the Room Shaper bottom chart, we can see that has been reduced by about -20 dB, so perceptually to ones ears, about one quarter as loud as it once was, at 10 Hz! I know of no other technology that can do this. Now that we have some objective proof that Room Shaper’s algorithm processing can be measured and audibly heard, what does it subjectively sound like on a variety of music? Subjective Listening The measurements and difference file above is with Room Shaper’s logarithmic Effect and Sense control set to maximum (i.e. default position). To my ears, it takes a little too much of the room resonances out of my living room. In my room the music sounds overly damped in the low end with the Effect/Sense controls set to maximum. With careful listening (at reference level 77 to 83 dB SPL) I found a balance of about halfway or 50% rotation of the Effect and Sense controls to provide the best balance. While adjusting the Effect control, I am listening for solid low frequency bass that did not activate the room, yet not sound overdamped. Then I adjusted the Sense control threshold to a level between being on all the time and just the peaks, while listening to music. Like the Effect control, there appears to be a sweet spot where the bass and lower mids sound crystal clear. Note there can be up to 1.5 seconds of delay between adjusting the Effect/Sense controls before hearing a change, including the bypass control. Remember this DSP software product works in the time domain. While I have posted this before, I use this list of tunes to evaluate gear as it has some decent dynamic range and I have heard most of this music over and over again for a dozen or so years on a wide range of audio gear. Sure, it is Dad rock 😉 To get into specific examples with some not from the list above: Excellent pop recording that is well produced/mixed. On the Power of Good-Bye, there are sustained low frequency bass notes that on my dual Rythmik 18” subs can pressurize my living room. With Room Shaper the bass notes are clear, articulate and deep with no overhang or room resonance. It is interesting to play with the effect control while listening in real time. It is almost like a focus control where the bass becomes the clearest sounding. Too little effect and the bass sounds resonant, too much sounds overdamped and losing bass. It takes a bit of time to train one’s ears to know what to listen for. The Room Mode calculator I mentioned in the preamble can assist in training ones ears. Remember this has little impact on the direct sound and is targeting beyond 250ms of room decay. As mentioned before, it is like blowing across a Coke bottle resonance where a particular tone or frequency takes a lot longer to decay than adjacent frequencies. This is what to listen for, but at low frequencies. And in ones listening environment, it is not just one frequency, but several frequencies (i.e. room modes), and each can potentially have its own decay time. Another difference I hear is the synth lower mids are much clearer. If you listen to the difference recording, one can hear a lower mid “peaky” resonant sound in my room. With that removed, one can hear the nice stereo effect of the keyboards with proper tone. The clarity is outstanding. One of my all-time favorite performances/recordings that I find really enjoyable to listen to is SRV’s Tin Pan Alley. I never get tired of hearing it and the fantastic use of dynamic range driven by Stevie and Double Trouble. The emotion of this blues tune goes up and down in volume over the course of the song to make one feel like you are there with the band, live at the venue. It is a crying shame there are so few “modern” rock, blues, pop, alt, recordings with this type of dynamic range that moves with the mood of the music. Unfortunately, extreme dynamic range compression has all but killed the musical life out of most recordings. In my room, Tin Pan Alley without Room Shaper has several bass guitar notes that resonate in my room. Switching in Room Shaper, not only tames those resonant bass notes, but provides a clearer picture of the lower mids, so that SRV’s guitar work sounds more articulate, as does the lower register of his voice in my room. I also notice the tone of the drums are clearer sounding without the room resonance masking the tone of the drums decay, especially the toms. I used this song to dial in Room Shaper’s Effect control. There is a sweet spot where the bottom end sounds tuned in, clear sounding with no overhang. It is like a focus control. Too little sounds muddy, too much sounds overdamped. I must say SRV has never sounded better on my system. As a guitar player, I am a fan of Eric Clapton. Learned to play some of his tunes. I like this particular live version of Badge. Excellent “live” recording. The bass guitar is turned up in the mix as is the kick drum and drums overall. Turning the volume up, the bass really pounds and one can feel the punch of the kick drum. For me, that’s the dream. I want to feel like I am at the concert or club listening to whatever band I feel like at the time with the punch and kick of a good live sound system. With dual 18” subs with 1800 watts and 4 x 15” woofers with 1000 watts covering 6 Hz to 630 Hz, I am getting there ☺ This song has great energy and just plain rocks out at the end with the bass thumping and drums pounding. You can feel the energy from the band and audience. For a stereo illusion, it is pretty convincing to close ones eyes and imagine being there. Lots of fun! And if you are a Cream fan, I can also recommend the Live at Royal Albert Hall 2005 video as well. Excellent recording/mix with a sense of being there. Looking at my listening notes, the dropping bass drum sound on Patricia Barber’s Regular Pleasures is a real treat as not only the impact of the drum sounds sharper, more impactful, it is being able to hear the decay of the drum sound clearer with Room Shaper on. Dire Straits, Six Blade Knife has the bass guitar level up in the mix and always sounded a bit “tubby” in my room, even with a flat frequency response in the low end. Room Shaper cleaned that right up being able to clearly distinguish between each note on the bass guitar with no room resonances. Stewart Copeland’s drums on Murder by Numbers by the Police sound like I am right in the drum room standing in front of the drums. The crack of the snare and punch of the kick drum sound incredibly tight and carrying through Stewart’s trademark articulation on each hit. The outro has the drum group cranked right up in the mix and bashes away. Love it. “Chaiyyaa Chaiyyaa” by Sukhwinder Singh & Sapna Awasthi has a variety of stereo effects that come through more clearly as does the drums and the drop bass as one goes further in the tune. Makes my wood house shake on its foundation, but in line with the rest of the decay times. Impressive! I could go on, but I think that by now one can see a consistent theme. Much clearer, non-resonant sounding bass and removal of “boxy” room sound. It becomes a new listening experience that was once only reserved for acoustically designed and constructed critical listening environments. Quite the innovation. Conclusion Indeed a new listening experience with Thierry’s Room Shaper product. Room resonances are real and measureable and to my ears, I can hear the difference readily with the addition of Room Shaper in the signal path in a positive way. A quick AB test with my lovely assistant switching Room Shaper on and off with my eyes closed, I could pick Room Shaper out 10/10 times. Once you hear resonant free low frequency sound reproduction and the “boxiness” of one’s room removed, it is hard to go back ☺ I am simply amazed that this can be accomplished using a software DSP VST plugin (Room Shaper plugin: 129€, 69€ for HAF service customers). Usually one needs to spend a small fortune on an acoustically designed and constructed critical listening environment. I say this as the last time I heard non-resonant low frequencies and an unboxy room was in an LEDE studio control room in Vancouver, Canada many years ago. Listening to the effect is like the room has disappeared. Or simply, the decay times across lower mid and bass frequencies are now similar in length. I find I can hear more into the music and less of my room. It is an audible, but subtle positive effect of making the low frequency sound clearer with no resonances and the added bonus of taking out the lower mid “boxy” sound of the room. One can hear the “boxy” sound on the keyboard synthesizers on the Madonna difference file. Very peaky buildup of lower mid sound being removed from my room. Closing my eyes, the sound field image is the size of the front wall. Not only do my fugly fridge size industrial speakers disappear (they did before with DRC), but the room is gone as well. No muddy resonant bass sound or room boxiness. Again, I am impressed the more I listen. Works a treat. A couple of operational notes. The plugin crashed (and JRiver exited) once while listening to music over an extended period (several hours). While the plugin works with video in JRiver, unlike convolution where the delay is known, there is no delay compensation with Room Shaper, so no lipsync. One may also need to increase audio driver and/or music player buffer sizes as lower millisecond values will have a tendency to stutter with drop outs. I did not hear any digital processing artefacts with Room Shaper engaged and as can be seen, it is limited to below 600 Hz frequency processing. For me, Room Shaper is the icing on the cake as I am already using SOTA DSP DRC tech for 3 way linear phase digital crossovers, driver time alignment, room frequency correction and excess phase correction. The ability to control low frequency room resonances and lower mid room characteristics is a new innovation. The ability to control how much is sensed and how much effect is applied (i.e. resonances removed) allows one to fine tune to personal preference. Once I found the sweet spot where the low end sounds focused with no resonances and the lower mids sound crystal clear, no further adjustments were required, no matter what music material was played. So it is set it and forget it. Very cool DSP innovation Thierry, congrats! Highly recommended. I hope you are enjoying the music! I wrote this book to provide the audio enthusiast with an easy-to-follow step-by-step guide for designing a custom digital filter that corrects the frequency and timing response of your loudspeakers in your listening environment, so that the music arriving at your ears matches as closely as possible to the content on the recording. Accurate Sound Reproduction using DSP. Click on Look Inside to review the table of contents and read the first few chapters for free. Mitch “Mitchco” Barnett. I love music and audio. I grew up with music around me, as my mom was a piano player (swing) and my dad was an audiophile (jazz). My hobby is building speakers, amps, preamps, etc., and I still DIY today. I mixed live sound for a variety of bands, which led to an opportunity to work full-time in a 24-track recording studio. Over 10 years, I recorded, mixed, and sometimes produced over 30 albums, plus numerous audio for video post productions in several recording studios in Western Canada.
  13. Thanks, but I also included music samples that one can compare to the original track and to another set of phones. Did you listen to those? No doubt audio is subjective, however, when it comes to measurements, some are pretty straight forward and indeed account for (very) audible differences between sound reproducers. For "accurate sound reproduction" if that is your goal means then the sound reproducer, whether headphones or speakers, should accurately track the input so the output is the same or as reasonably close to it as possible. So when it comes to frequency response, which accounts for a large part of what we hear and judge sound reproducers on, a smooth frequency response is a basic requirement. Unfortunately, both measured and in the recordings, the LCD-4z does not meet this basic requirement in the top end. It is not smooth. While one's preference may differ, the LCD-4z's are not accurate sound reproducers. Wrt speakers, Floyd Toole and Sean Olive with many years of research has shown that a speaker that measures well on and off axis in an anechoic environment will indeed subjectively sound good in a normal listening environment. In fact, their results show a high correlation (86%) between their objective measurements and controlled subjective listening tests with many test participants. Their predictive model in estimating the in-room frequency response based on "spinorama" anechoic measurements is so good it is included in the CTA 2034 A Standard Method for Measurement In Home Loudspeakers that you can download for free. Look at Fig 11 on page 37 that shows the predicted in-room response based on anechoic measurements compared to the actual measured in-room frequency response - they are identical. Olive did the same work with headphones with hundreds of participates over several years, which shows most everyone has similar preferences when it comes to what makes for a good sounding headphone. Hence the Harman Target Curve. Those are linked in my objective review. While I agree that audio is subjective, there is considerable research into correlating objective measurements with subjective listener preferences, to the point where industry standards are developed from this research. Let's not throw out the baby with the bathwater and perhaps raise the bar a little to what constitutes good sound, which can be objectively measured. Here is yet another data point using Harman's test rig on the LCD-4 (not Z) but has the same issues my measurements show with non smooth high frequency response. The next measurement is the calculated score based on deviation from Harman Target Curve using their predictive model. Score is 65%. It suffers from having 5-6 dB less bass and treble.. which you can see by looking at the RED error target curve and with the non smooth high frequency response.
  14. Sorry @Facel I cannot disclose at this time, but the review will be up soon...
  15. @Facel Thanks for asking. No, I have not as I am still adding to the DSP software product portfolio I intend to cover in the next edition. Even though the DSP concepts and most of the procedures in the current book apply to virtually all digital loudspeaker and room correction products, I am getting into the implementation details of each DSP product. I am also adding a chapter on integrating subwoofer(s) as this is a problematic area, especially if one wants to time align subs with mains. There has been new research into target frequency responses correlating to listener preferences, and a new industry standard on measuring loudspeakers and predicting in room responses that are being added to the book. Right now, a new and interesting DSP software product is being added and there will be a review of it on AS shortly. Sorry, I don't have an ETA on when the next edition with be available, but I am working on it. Kind regards, Mitch
  16. Agree with mansr. I have been doing it for years into a 3 way active system with no loud mishaps. In my case I use JRiver's internal volume control which also has volume protection. Not sure if foobar2000 has that or if it even really matters. As mansr noted, just need to be a bit careful is all.
  17. Hi @omid Very nice! If you are getting a good step and frequency response with REW, then everything is fine. Does the Xonar card come with an ASIO driver? One thing you could try if you have not already is to check the response of the signal path sans speakers and mic. But if you are getting a good step and frequency response out of the speakers, then it is likely there are no issues at all. Wrt the attenuation of the digital filter, in JRiver's convolution you could click on normalize filter volume which would bring some gain back. Is there a similar setting in Roon? Another way is to add a digital gain stage. In JRiver, one could use the eq plugin just to add gain, but with no eq. There are VST plugins that can do this as well. I have used Blue Cat's Gain Suite to do this. Just have to watch for any clipping. Seems like you have it all sorted. I think Bernt is working on some of the multichannel issues in AL V6, so this may also allow the use of Wasapi or ASIO. But it seems to me there is nothing to worry about. Cheers, Mitch
  18. @detlev24 Thanks for your kind words and comments! Given I like large, high efficiency loudspeakers with horns, I like your reference choice 🙂 Re: Generally, I would like all manufacturers to be more transparent and more accurate with the data they release! Totally agreed! Especially since there is a standard for loudspeakers measurements called, "CTA-2034-A Standard Method of Measurement for In-Home Loudspeakers". The standard is publicly available and free to everyone. "This standard describes an improved method for measuring and reporting the performance of a loudspeaker in a manner that should help consumers better understand the performance of the loudspeaker and convey a reasonably good representation of how it may sound in a room based on its off-axis response and how this response affects the consumer’s experience." It is well worth the read and embodies the accumulation of work by Floyd Toole, Sean Olive and others in a industry acknowledged standard. It offers a standard report format that consumers should demand to see for any given speaker purchase. Sometimes referred to as "spinorama" report. The only place I know that informally catalogs similar reports is: https://speakerdata2034.blogspot.com But given the science of designing and measuring loudspeakers, that has been proven over and over again, even since Floyd's spinorama paper from 2002, it seems very few speaker manufacturers have cottoned onto this. I hope consumers put more demand on loudspeaker manufacturers to produce these reports, especially the new CTA 2034 A standard (from 2015 to be updated by the end of this year) format where it is clear to see the speakers on and off axis response and especially the predicted in-room frequency response. Cheers!
  19. Hi @omid Welcome aboard! And thanks! Perhaps you can briefly describe your measurement mic, gear and signal path for taking measurements? I doubt you are missing anything, but always good to have a 2nd pair of eyes have a look. Having said that, if you are happy with the SQ, then stick with what works. But lets have look anyway 🙂
  20. In this objective review of the Audeze LCD-4z headphones, I measure the frequency response and provide binaural recordings of the headphones so that folks can download and listen to music samples. The music samples are compared to the original track so one can hear, relatively speaking, any differences between the source and headphone reproduction. In addition, there is a comparison to the NAD Viso HP50 headphones both from a frequency response and binaural recording perspective. The binaural music recording switches back and forth every 10 seconds between the two headphones, so one can evaluate tonal differences between the two. Preamble A few points to put this review into context. Unlike most headphone measurement rigs that use a dummy head, I am using my own noggin and ears as the dummy head (still a dummy head says my wife). This way I can adjust the fit and feel of the headphones to sound just right to my ears. Many headphone measurement rigs require a “compensation curve” to interpret the raw headphone measurements. Tyll does a good job of explaining that in the link above. However, in this particular case, I have found a set of in-ear measurement mic’s that seem to not require (much of) a compensation curve, up to 10 kHz anyway. More on that point later. Just like loudspeakers, there are subjective listening tests correlating to objective measurements to indicate that there is a “target curve” for headphones that correlates to what sounds neutral or preferred or accurate. Sometimes referred to as the Harman target curve. Tyll, in the link above, discusses it and I wrote a summary, “The Science of Preferred Frequency Response for Headphones and Loudspeakers” peer reviewed by Sean Olive. There is an educational presentation by Sean and team on, “The Perception and Measurement of Headphone Sound Quality: Do Listeners Agree on What Makes A Headphone Sound Good?” It is interesting to read about the controlled subjective listening tests, objective measurements and conclusions. Since then, the subjective listening tests have been repeated several times, with hundreds of participants, arriving at the same objective results and encapsulated in a more recent 132 page presentation. Regardless of headphone compensation curve, I am using the NAD Viso HP50 headphones as a comparison point with the LCD-4z’s. The reason why HP50 is because they sound and measure the most neutral compared to a reasonably large sample size of headphones. From the link above, Tyll’s comment about the HP50’s (and Focal Spirit Professional), “These two headphones are among the most neutral I've heard, and they do match the Harman target response quite well relative to other headphones I've measured.” At the time Tyll has measured and reviewed some 200 headphones. Having measured and reviewed the HP50 myself, I achieved similar measurement results and agree with Tyll’s subjective statements, quite a neutral response, albeit down in level in the top octave. Using my measurement setup, whatever the frequency response curve measures out for the HP50 can be considered, relatively speaking, neutral or a point of reference for comparison. Then by measuring the LCD-4z, we can compare to “neutral”. This will correlate with the binaural recordings as well. Meaning, with the binaural comparisons of the Audeze LCD-4z to the source track and to the NAD HP50 headphones (that measure and sound neutral), we can characterize the tonal response of the LCD 4z’s relative to an ideal or neutral reference. On a technical note, the Audeze LCD-4z headphones have a rated impedance of 15 ohms. This means many sources can drive these headphones to satisfying levels without the need for a separate headphone amplifier. However, with this low of an impedance, one needs to be careful that the source device driving the headphones should have, as a rule of thumb, an output impedance at least 1/8th the impedance of the headphones. So 1/8th of 15 ohms, is less than 2 ohms. Therefore, the output impedance of the device used to drive these headphones should be less than 2 ohms. My Lynx Hilo headphone’s output impedance measures about 0.08 ohms, so no problems. Here is a link to several headphone devices output impedance that one can compare. At the end of the article I include an appendix on the calibration and verification of the test gear that I used to measure frequency response and make the binaural recordings. In a nutshell, the test equipment has flat frequency and phase response with low distortion. I.e. a transparent recording chain. With the preamble out of the way, let’s move on to the objective measurements. Objective Measurements If you don’t want to look at graphs and would rather listen to the binaural recordings, click here. Here is my methodology to reduce variation in getting a good headphone seal and consistent frequency response measurements. I took ten measures or five stereo pairs, and for each measurement, I adjusted the binaural mics and the headphones to give the best and most consistent frequency response. Here is the left and right channel frequency response for one pair of measurements using REW: As one can see, an incredibly flat frequency response from 20 Hz to 1.6 kHz, with a -3 to -4 dB dip between 1.6 kHz and 4.1 kHz. Then a +5 to +8 dB peak at 5.3 kHz. From there, a -20 dB dip at 7.8 kHz and then a +16 to +18 dB peak at 9.9 kHz. Then another -20 dB dip at 14 kHz. Getting a good seal with these headphones is no problem, but a bit of a bumpy ride in the treble. Now some of that is due to timbral issues with head related transfer function (HRTF) and that it is my head. It is also related to binaural recording techniques whether using the blocked ear method or microphones at the ear drums. Further, getting the binaural mics into one’s ears in the same place each time has variability, as does the fit and positon of the headphones. In the case above, even variability at the top end between the left and right channels due to the right mic’s position. It isn’t easy trying to reduce these variables to record realistic headphone measurements that are repeatable. But close enough to be the same trend in tonal response to match, which is what we are after. One can also compare to other measurements, as we will with the NAD Viso HP50 and correlate the differences, not only in the measurements, but in the binaural recordings as well. Let’s look at a distortion measurement. I would not put too much emphasis on the absolute distortion levels as the test was performed at regular listening level, not to a measurement spec. Also, I have no idea how much distortion is being contributed by the microphones as there is no distortion spec for them. What we are interested in is the overall spectral shape of the distortion versus frequency to get an idea of where there might be some potentially audible issues. Rising distortion in the low end is typical of any headphone or loudspeaker for that matter, often in large numbers, but reasonably low for the LCD-4z: There is a rise in distortion, relatively speaking, from 3.5 kHz to 8.5 kHz. For those wondering why the distortion measures only goes up to half the sample rate can read the REW help. Let’s bring in the NAD HP50’s so that we can make a measurement comparison. I matched the sound pressure level for both headphones by adjusting volume using pink noise as the source through REW so that they matched in level. This was both for the measurements and binaural recordings. I followed the same procedure as mentioned at the top of the objective measurements. HP50 left and right channels – no smoothing: Here we can see an extended response down to below 20 Hz and mostly flat out to 1 kHz with a bit of dip from 1 kHz to 3.5 kHz and the then a rise and only after 10 kHz we see a roll off with some peaking and dipping. Note it is much smoother in the treble than the LCD-4z relative to the magnitude of the peaks and dips. The one thing to notice is that I could not get the best seal on the right ear cup and shows as a channel imbalance in the low end. Don’t really notice it when listening to music. The HP50’s have small ear cups and for my large ears, with glasses and long hair, make it next to impossible to get a good seal. They did at the beginning, but that was due to the brain crushing clamping force applied until I nearly broke the darn things stretching them out. Again, a little bit of right channel imbalance at the top as a measurement artifact due to mic positioning. But comparing to my previous measurements of the HP50, I can verify consistency and repeatability: The red and blue traces are from this review and the green and gold traces are from my previous review of the HP50’s from 1 ½ years ago. As we can see in the previous review, aside from the small low end seal differences, the left and right channels track closely to about 10 kHz. This is why I know that the blue curve in this review is with the right mic not quite positioned correctly. But the left earphone from this review tracks very well with both left and right channels from the previous measurements. As can be seen, I have a little work to do on getting both mics in the same place for left and right measurement consistency. A distortion plot: Pretty smooth with rising distortion in the low end, which our ears are insensitive to, a narrow spike at 1.93 kHz and some rising distortion between 4 kHz and 6 kHz with the right earphone a little higher than the left. Now let’s look at some comparisons so we can make predications on what we think we are going to hear in the binaural recordings. Here I have overlaid the LCD-4z and the HP50’s frequency response – both channels, no smoothing: Both headphones, from a tonal perspective are very close from 20 Hz to 700 Hz, with the HP50’s having a bit more output in that range. The HP50’s are up to -3 dB down from 1 to 1.5 kHz and then it is a bit of a mess to try and sort out after that. To make it easier, I have overlaid the individual left and right frequency responses for comparison. First the left side: As previously mentioned with the HP50 (green trace) having a bit more low and mid frequency output, both have a pretty smooth response up to 6.5 kHz. We see the LCD-4z (blue trace) have a -3 dB drop from 1.5 kHz to 4.5 kHz relative to the HP50’s. There is a -12 dB drop at 7 kHz relative to the HP50’s. Conversely, we see about a +10 dB spike at 9.3 kHz for the LCD-4z relative to the HP50. At the very top end, we see that the LCD-4z has +22 dB more output that 18.3 kHz than the HP50’s. Looking at the right earphone: Good tonal match out to 6.3 kHz. Relative to the HP50’s there is about -10 dB drop in the LCD-4z at 7.6 kHz. At 10 kHz, the LCD-4z have about +10 dB more output relative to the HP50’s at 11.6 kHz and the LCD-4z’s have about +16 dB more output than the HP50’s at 17.8 kHz. Note these graphs are not smoothed. Meaning they are full resolution measurements, but it is commonly acknowledged that our ears don’t hear that narrow bandwidth of frequency resolution. Our ears hear about 1/6 to 1/12 octave bandwidth. So if I apply 1/6 octave smoothing, we can get a better handle on the tonal differences that we expect to hear relative to the two headsets as compared to each other and to the original source music track. Rather than repeating the same graphs above with the smoothing, let’s just look at the left channel between the two headphones. I feel this best represents each headphones sonic signature from a measurement perspective: The HP50 (green trace) have about 4 dB more output in the low frequencies and lowers mids than the LCD-4 from 20 Hz to 750 Hz. Relative to the HP50, the LCD-4z has up to 2 dB more output from 800 Hz to 1.5 kHz. The LCD-4z’s have about -3 to -5 dB less level from 1.5 kHz to 4 kHz as compared to the HP50’s. LCD-4z are -7 dB down at 7.4 kHz relative to the HP50. LCD-4z +9 dB at 9 kHz. LCD-4z +8 dB at 15.4 kHz LCD-4z +21 dB at 18.7 kHz. Comparatively speaking, I expect the HP50’s to have a bit more bass and lower mid output than the LCD-4z’s, both exhibiting a nice smooth response (LCD-4z ruler flat!) with extension to below 20 Hz. Might not be much of an audible difference in the 800 Hz to 1.5 kHz range, but could make the voice on the HP50’s not stand out as much in the midrange. The LCD-4z’s will likely sound recessed in the 1.5 kHz to 4 kHz range, which is also the range our hearing is the most sensitive. So not as bright sounding in the upper mids as compared to the HP50’s. That plus the LCD-4z with the peaks and valleys in the treble response, and the LCD-4z’s having (much) more high frequency output than the HP50’s past 13 kHz will be the most audible differences, I predict. Distortion plot of both left channels: The HP50’s have a bit more distortion in the low frequencies and the LCD-4z more distortion in the 4 kHz to 11 kHz range. Seems to be tied more to the magnitude response level differences... I am not convinced at what level our ears can hear THD as they are frequency dependent. It also requires calibration with a set of mics that have known distortion specs, which these binaural mics do not. So how much distortion is from the mics versus how much from the headphones? And how much is really audible? Not sure if this is providing value as our ears seem to be (far) more sensitive to frequency response variations than anything else. Want to hear it for yourself? Try ABX Testing and Distortion. Let’s listen to see if these frequency response differences are audible in the binaural recordings. Subjective Listening using Binaural Recordings OK here is where the fun begins. I recorded a snippet from Tracey Chapman’s Fast Car. It is a familiar recording with good bass and dynamic range with her voice mixed up in level so it stands out and makes it easy to hear any tonal imbalances. Also pay attention to the “s” in her voice as it can range from dull sounding to overly sibilant. I am using a portion of the song on the basis of fair use for the purpose of evaluation and research. I am not doing this for financial gain. Please delete these music samples once you are done with the evaluation. I level matched the binaural recording to the original track in my DAW (Mixcraft) and then using the DAW’s automation, synchronize the two stereo tracks and switch back and forth every ten seconds so you are getting the benefit of instant switching between the source music track and the binaural recorded track. Looks like this in Mixcraft: There are some “left in” edits when the tracks switch. These can be used as an audible cue to focus in on the new sound to assess the differences before you get used to it as we quickly adapt. I picked a random spot, just before Tracy sings and then a couple of choruses. We start off with the original track for ten seconds and then switch to the binaural track of the Audeze LCD-4z, then back to the original, switching back and forth every ten seconds until the end of the music selection. Tracy Chapman Fast Car Audeze LCD-4z (WAV File - 33.1 MB) (ZIP File - 29.9 MB) To my ears, the bass and mids sound pretty close together, but the upper midrange and treble sound different to my ears. The tonal response of the LCD-4z sound like their measured frequency response where some frequencies are missing in Tracey’s voice, and other frequencies are too much, giving it a distinctive tone and a little too bright sounding overall in the top octave. Overly sibilant sounding as compared to the source is what I hear. While this may be some folks preference, not that there is anything wrong with that, it does deviate from a neutral response like the ruler flat bass and mids that the LCD 4z’s do have. There is some context here as well. The headphones are recorded using my ears with a certain HRTF and is going to have “some” impact. But as you listen, when it is a match, like in the bass and mids, the source and the binaural recording sound amazingly similar. I would love to hear your feedback on what you hear. Are you hearing similar to what I am hearing or different? Now let’s take a listen to the source track and compare with the NAD Viso HP50. Again, starts with the source track and then in ten seconds switches to the HP50, then ten seconds later, back to the source, switching back and forth every ten seconds to the end of the music selection: Tracy Chapman Fast Car NAD HP50 (WAV File - 32.6 MB) (ZIP File - 29.3 MB) To my ears, again the bass and mids sound very similar to the source. Also, Tracy’s voice sounds similar to the source from a tonal perspective. It does not sound “filtered.” However, as seen in the frequency response measurement, the top octave after 10 kHz drops off fairly quickly as the attack on the guitar strings, the hi-hat/cymbals and Tracy’s “s” all sound down in level (i.e. dull) compared to the source track. A little shelving eq or treble control boost above 10 kHz would bring the level back up. Then it would be more difficult to tell the two apart. There are balance issues between left and right channels that one can hear in this binaural recoding (and the LCD-4z too) as the center image shifts a bit when switching. This is due to the binaural mics not sitting perfectly in each ear canal. What are you hearing? Finally, a direct comparison of the LCD-4z to the HP50’s: Tracy Chapman Fast Car LCD4z and HP50 (WAV File - 32.9 MB) (ZIP File - 29.5 MB) Regardless if listening over headphones (preferred) or speakers, and regardless of individual HRTF’s, it is a direct comparison where one can hear the “relative” differences. The two big differences that stand out for me is that Tracy’s voice sounds close to as it should with the HP50’s whereas listening to the LCD-4z’s sounds like the upper midrange of the frequency range is missing in her voice and then accentuated too much in the treble (listen to the “s’s”). The treble response in the HP50 is rolled off, but easy to fix with a basic high frequency (shelving) tone control. Not so easy with the LCD-4z as it is going to require parametric eq to deal with the ups and downs in treble frequency response. What are you hearing as the differences between the two headsets? Conclusion The Audeze LCD-4z have an incredible flat frequency response in the bass, with good extension below 20 Hz and flat midrange response up to 1.5 kHz. This is evidenced by the measurements and correlated with the binaural recording comparing to the source track. I mean it sounds and measured perfectly flat with excellent matching channel balance. Also matches the Harman target curve for a neutral frequency response pretty well up to 1.5 kHz. From there, we get into deviations from neutral with a little lower level output in the 2 to 4 kHz range, which is right in our human ears most sensitive range. Then we get into some peaks and dips in the upper midrange and treble. Not only measured, but also heard on the binaural recordings, both comparing to the original source track and the NAD HP50’s. Finally, overall a little too much high frequency energy past 13 kHz, both measured and heard. Of course there are complicating factors related to measuring headphones and making binaural recordings using a person’s (or dummy) ears (mics fitting properly and repeatedly is one example). While it may not represent the absolute truth, from a comparative perspective, it appears to be quite accurate as there is not only good correlation between the measurements, but also comparing to the source music track. The direct comparison differences between the two headphones should hold true, regardless of one’s own HRTF, headphone or speaker brand while listening to the binaural recording. As a side note, I believe this binaural recording technique is sensitive enough and enough resolution that if there is an audible difference when directly comparing two devices, using music as the test signal, then the recording will pick it up and accurately represent it on playback. One can always eq the LCD-4z’s towards a neutral response if desired. The data to implement parametric eq (PEQ) is contained in this review where inverse filters for the peaks and dips could be created for both channels. Bring up the response in the 1.5 kHz to 4 kHz range and then bring the overall treble response down to neutral (or taste) and. I would use the 1/6th octave smoothing graphs as the guide for PEQ. Or if you really want to fully optimize the response, one could get a pair of binaural mics, a mic preamp and use DSP software like Audiolense or Acourate to accurately tailor the response to be neutral, and matched to your own HRTF. Or even try the Impulcifer which is like an open source version of the Smyth Realiser. I did try Audeze Reveal, which has an eq preset for the LCD-4z, as a plugin in JRiver. To my ears, it was a subtle difference and did not tonally change or reduce the treble response. I did not get a chance to see how customizable Reveal is. Finally, make sure the device you are using to drive the LCD-4z follows the 1/8th rule for impedance matching. For a 15 ohm headset, the driving device needs to have less than 2 ohms output impedance. I hope you are enjoying the music! Appendix: Calibration and Verification of Measurement Equipment The purpose of this section is to list the test equipment I used and verification measurements performed to ensure a transparent signal path so as not to influence the headphone measurements or binaural recordings of the headphones. Test sample rate is 48 kHz and all digital and analog levels on the Lynx Hilo are set to 0 dB unless otherwise noted. Sound Professionals MS-TFB-2 Binaural Microphone I have tried many binaural microphones and dummy head set ups over the years. I find the Sound Professionals MS-TFB-2 to be a unique set of binaural mics as when put in-ear, they have an incredibly flat frequency response. No other binaural mics I have tried have this characteristic. Mine were ordered with the XLR ends and phantom power converter so they can be used with a mic preamp that has phantom power. You can read the rest of the specs from the link above. Contacting the supplier who indicates the frequency response was measured in an anechoic chamber using the standard for measuring microphones. While I can’t verify the frequency response, I can say over the years using this mic set for a number of binaural recordings of music and nature events plus audio gear, I am reasonably confident of the spec. A shout out to Chris at The Sound Professionals for answering my many questions in a timely manner. ART PRO MPA II Microphone Preamplifier The ART PRO MPA II is a stereo microphone tube preamplifier that in a shoot-out out ranked several top name brands. Using REW, I routed the digital test signal through the Lynx Hilo DAC analog output and input to the mic preamp with the mic preamp’s output feeding into the Lynx Hilo’s line input ADC and routed back to REW for display. For using a studio pre as a measurement preamp, it did pretty well: -1 dB down at 15 Hz and ruler flat to over 20 kHz using a 48 kHz sample rate. I did run a frequency response test at 192 kHz sample rate and the preamps response was -1 dB down at 50 kHz. Looking at Phase: Flat phase response. Ah, we see the right channel with a higher level of distortion than the left. The good news is that it is linear across the frequency range and at 0.02 percent distortion, very likely inaudible. Btw, that’s with 20 dB of microphone preamp gain applied. What causes the small channel imbalance? It’s just tubes man. I should also point out, these are the same settings (i.e. 20 dB of preamp gain) when making the binaural recordings. The binaural recordings may not be the most quiet, not only due to my environment, but I was quite conservative with headroom. I could have achieved another 10 dB or even 20 dB at best signal to noise ratio, but in the context of “comparative recordings”, this should not cause an audible issue. One should realize that when measuring the mic preamp, I am also measuring the Lynx Hilo AD and DA converters as well, using the balanced XLR connections. Measuring just the Hilo’s analog loopback using balanced connections puts the THD distortion below 0.0005 percent. But since I am using the headphone output, let’s see the Hilo’s headphone amp measurements in an analog loopback test. Lynx Hilo AD DA Converter The Hilo still measures at the top of a long list of pro interfaces for AD/DA linearity and transparency. Here I am measuring the “analog” loopback of the converter through the headphone amp. Meaning I have taken the headphone amplifier unbalanced analog output and fed it back into the Hilo’s analog line input. Again using REW as the measurement software, we see a straight pass through with minimal distortion. Frequency response. Phase response. Distortion. Highest THD was at 30 Hz at 0.0043 percent as per cursor. I could get it to go a bit lower if I moved the connection cable around a bit, so I suspect that this is more related to low frequency noise with an unbalanced connection than distortion. Nonetheless, below our audibility thresholds. Conclusion – the measurement and recording chain verified with a flat frequency and phase response and low distortion. I won’t be including these measurements in future articles, other than sharing a link to these verification measurements. I wrote this book to provide the audio enthusiast with an easy-to-follow step-by-step guide for designing a custom digital filter that corrects the frequency and timing response of your loudspeakers in your listening environment, so that the music arriving at your ears matches as closely as possible to the content on the recording. Accurate Sound Reproduction using DSP. Click on Look Inside to review the table of contents and read the first few chapters for free. Mitch “Mitchco” Barnett. I love music and audio. I grew up with music around me, as my mom was a piano player (swing) and my dad was an audiophile (jazz). My hobby is building speakers, amps, preamps, etc., and I still DIY today. I mixed live sound for a variety of bands, which led to an opportunity to work full-time in a 24-track recording studio. Over 10 years, I recorded, mixed, and sometimes produced over 30 albums, plus numerous audio for video post productions in several recording studios in Western Canada.
  21. Not sure how to answer that as I am following industry guidelines for monitoring music production. A couple of examples are: EBU Tech 3276-1998 Listening conditions for sound programme material and Recommendation ITU-R BS.1116-3 (02/2015) Methods for the subjective assessment of small impairments in audio systems. There are others, but other than the frequency response target and that the monitors are not elevated to accommodate a control room window to the studio, this is how most music is recorded, mixed and mastered, using these guidelines for room setup. The guidelines should be updated based on the work of Sean Olive and Floyd Toole have done showing peoples preference for a more neutral response as per one of my posts above. So if the goal is to reproduce as closely as possible to what was heard in the studio (or venue or concert hall or whatever) by the artists, producers and engineers, I would follow the guidelines. If not, then anything goes 🙂 There is no right or wrong. Every system I have set up or worked on, whether pro or home, I use the equilateral triangle and follow the guidelines. The soundstage always sounds right to me, with a solid phantom center. But that's my preference. I can't listen to speakers that point straight ahead or the triangle is too wide or narrow or the LP is too close or too far from the speakers. In all of those cases, the soundstage does not sound right to me. But again, that's my personal preference to align with industry guidelines. For others, likely different and in some case very different, but I would not argue with anyone's preference if it is different than mine. I don't know the details of your setup to say one way or another, other than to say there is an industry "best practice" that I follow,. It may not be your preference.
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