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Como a página virou...
Material relevante, postado pelo Bog em outro tópico:
Ele disponibiliza o áudio dos teste em alta qualidade, além de outros artefatos, como plugins usados e quase tudo o que foi dito, no Word (script) aqui:
Transcrição do vídeo, incluindo a minutagem:
"Audio myths, artifact audibility, and comb filtering—understanding what really matters with audio reproduction and what does not - AES Workshop was presented on October 12, 2009
Workshop Chair: Ethan Winer, RealTraps - New Milford, CT
Featuring: James "JJ" Johnston, DTS, Inc. - Kirkland, WA; Poppy Crum, Johns Hopkins School of Medicine - Baltimore, MD; Jason Bradley, Intel Corporation - Hillsboro, CO
PART 1 – Excerpts from live show
00:00 [Slide of AES web page]: Hi, I'm Ethan Winer, and this is a video version of my AES workshop presentation from October 12, 2009. At that workshop I was joined by hearing experts James Johnston and Poppy Crum, who each spoke for about half an hour. For length and other considerations, only a small portion of each speech is repeated here.
[Flash www.ethanwiner.com/aes on screen] Many of my demonstrations include audio examples. However, YouTube re-compresses the audio, so you can download the original full-quality Wave files from my web site.
00:31 ETHAN WINER INTRODUCES THE PANEL
1:05 JAMES JOHNSTON explains "Why do things always sound different?" Show slides 14-16 plus parts of his video.
5:20 POPPY CRUM does her presentation.
9:36 ETHAN WINER presents the rest below.
9:52 EYEWITNESS VIDEO [Play the video up to 1:51]
11:57 MIX ENGINEER: Anyone who records and mixes professionally has done this at least once in their career—you tweak a snare or vocal track to perfection only to discover later that the EQ was bypassed the whole time. Or you were tweaking a different track. And if you've been mixing and playing around with ... whether you're a professional or just a hobbyist, if you've been doing this for a few years and you haven't done that, then you're lying. Yet you were certain you heard a change! Human auditory memory and perception are extremely fragile, and expectation bias and placebo effect are much stronger than people care to admit.
[JJ injects, EW comments about the "producer's" channel strip.]
And these are the points of course that ... Some of you know me from the web forums where Jason and I are both active, but ...
The result is endless arguments over basic scientific principles that have been understood fully for more than fifty years—the value of ultra-high sample rates and bit depths, the importance of dither and clock jitter, and even believing that replacement AC power cables can affect the sound passing through the connected devices. An entire industry has emerged to sell placebo "tweaks" to an unsuspecting public. Let's look at some of the more outrageous examples:
13:21 OUTRAGEOUS AUDIOPHILE NONSENSE SLIDES: (Improvised, no script.) Brilliant Pebbles, more Brilliant Pebbles, Quantum Clips, Acoustic ART Resonators, Marigo Dots, ESP Music Cord, Furutech DeMag.
16:49 PART 2 – Ethan's Presentation
WTF?: How do companies convince otherwise sane people to pay $129 for a jar of rocks? Or $3,000 for magic bowls way too small to possibly affect acoustics? Or thousands of dollars for a replacement power cable? There are even "audiophile grade" USB cables costing hundreds of dollars. More important, why do people think they hear a difference—always an improvement, of course!—with such products?
17:18 ACOUSTIC COMB FILTERING: Through my research in room acoustics I believe the acoustic phenomena known as comb filtering is one plausible explanation for many of the differences people claim to hear from cables, power conditioners, mechanical isolation devices, low-jitter external clocks, ultra-high sample rates, replacement power cords and fuses, and so forth.
17:37 [Slide 1: 18" away from the wall] Comb filtering is a specific type of frequency response error that occurs when direct sound from the loudspeakers combines in the air with reflections off the walls, floor, ceiling, and other nearby objects. This graph shows the response I measured 18 inches away from a reflecting sheet rock wall.
17: 55 [Slide 2: With and without RFZ absorbers] In this graph and the previous one, you can see the repeating pattern of equally spaced peaks and deep nulls. The peak and null frequencies are related to the delay time, which in turn is related to the distance of the reflecting surfaces. This particular graph compares the response measured with and without absorption at the side-wall reflection points in my living room.
18:15 [Slide 3: Reflections off a wall colliding] Peaks and deep nulls occur at predictable quarter-wavelength distances, and at higher frequencies it takes very little distance to go from a peak to a null. For example, at 7 KHz a quarter wavelength is less than half an inch! At these higher frequencies, reflections from a nearby coffee table or even a leather seat back can make a big change in the frequency response at your ears.
18:35 [Slide 4: Lab room measurements taken four inches apart] Because of comb filtering due to multiple reflections in a room, moving even a tiny distance changes the response a lot. Especially in small rooms having no acoustic treatment. The response at any given cubic inch location in a room is the sum of the direct sound from the speakers, plus many competing reflections all arriving from different directions. This graph shows the frequency response for two locations in the same room only four inches apart. Yet it looks like different speakers in a totally different room!
19:04 LOUDSPEAKER DISTORTION: Keeping what truly matters in perspective, it makes little sense to obsess over microscopic amounts of distortion in an A/D converter when most loudspeakers have at least ten times more distortion. This graph shows the first five individual components measured from a loudspeaker playing a 50 Hz tone. When you add them up the total THD is 6.14 percent, and this doesn't include the IM products we'd also have if there were two or more source frequencies.
19:31 ROOM RESPONSE: Likewise, compared to even very modest gear, all domestic size rooms have huge variations in low frequency response, comb filtering from untreated reflections, and substantial ringing at a dozen or more modal frequencies. This graph shows the low frequency response at high resolution as measured in a bedroom sized space. Does it make sense to obsess over "gear" when listening environments are by comparison so much worse? As audio professionals we should strive for the highest quality possible. Of course! But it's important to keep things in perspective and be practical. My intent is not to belabor the importance of acoustics, but to put in perspective what parts of a playback system do the most damage. Most sensible people will aim to improve the weakest link first.
20:15 AUDIO PRECISION TESTER: There's also anti-science bias by those who believe specs don't matter, and "science" doesn't know how to measure what they are certain they can hear. If it weren't for science, we'd all be banging on tree stumps in a dark cave. As JJ explained, every time you play a recording it sounds a little different. Further, if you move your head even an inch or two, the frequency response can change substantially due to acoustic comb filtering, especially in an untreated room. And the more you hear a piece of music, the more likely you'll notice details previously missed. Is that triangle hit clearer because you recently added a power conditioner, or simply because you never noticed it playing before? Understanding that test gear is far more reliable and repeatable than human hearing is the last frontier in stamping out audio myths.
21:00 HEY, IT'S YOUR MONEY: Ultimately, these are consumerist issues, and I accept that people have a right to spend their money however they choose. I am not opposed to paying more for real value! Parts and build quality, features, reliability, and even appearance cost more. For example, some DI boxes cost $30 and others cost ten times more. If I'm an engineer at Universal Studios recording movie scores, which can cost hundreds of dollars per minute just for the musicians, I will not buy cheap junk that might break at the worst time. My only aim here is to explain what affects audio fidelity, to what degree of audibility, and why.
21:35 FOUR PARAMETERS: The following four parameters define everything needed to assess high quality audio reproduction:
• Frequency Response
• Time-Based Errors
There are subsets of these parameters. For example, under Distortion there's harmonic distortion, IM distortion, and digital aliasing. Noise encompasses tape hiss, hum and buzz, vinyl crackles, and cable "handling" noise known as the triboelectric effect. Time-based errors are wow and flutter from vinyl records and tape respectively, and jitter in digital systems.
22:10 Aside from devices that intentionally add "color" by changing the frequency response or adding distortion, it's generally accepted that audio gear should aim to be transparent. This is easily tested by measuring the above four parameters with various test signals. If the frequency response is flat to less than 1/10th dB from 20 Hz to 20 KHz, and the sum of all noise and distortion is at least 100 dB below the music, a device can be said to be audibly transparent. A device that's transparent will sound the same as every other transparent device, whether a microphone preamp or DAW summing algorithm.
22:45 Of course, transparency is not the only goal of audio gear. Euphonic distortion can be useful as "glue," but there's no need for magic. Transformers can add distortion. Tubes can add distortion. Tape distorts if you record at high levels. But do we really need to spend thousands of dollars on boutique gear to get these effects? Are there other, more practical and affordable ways to get the same or similar results? Regardless, it is impossible to argue about the subjective value of gear "color," so I won't even try.
23:13 ALL THE DATA PLEASE: Although product specs can indeed tell us everything we need to know, many specs are incomplete, misleading, and sometimes even fraudulent. But this doesn't mean specs cannot tell us everything that's needed to assess transparency—we just need all of the data. Common techniques to mislead include using third-octave averaging for microphone and loudspeaker response, and specifying a frequency response but with no plus or minus dB range. Or using very large divisions, like 10 or 20 dB each, to make a ragged response look more flat.
23:43 [Slide 1: True loudspeaker response] I measured this loudspeaker from about a foot away in a fairly large room. This graph shows the true response as measured, with no averaging.
23:51 [Slide 2: Same loudspeaker response averaged] This graph shows the exact same data, but with third-octave averaging applied.
23:56 [Slide 3: Same response again with 20 dB per division] This graph shows the same averaged data again, but at 20 dB per vertical division to make the loudspeaker look flatter than it really is. Which version looks more like what speaker makers publish?
24:07 ARTIFACT AUDIBILITY: Masking is a well-known principle by which a loud sound can hide a softer sound if both sounds have similar frequencies. This means you can hear treble-heavy tape hiss more readily during a bass solo than during a drum solo. Cymbals and violins contain a lot of treble, so that tends to hide hiss. The masking effect makes it difficult to hear artifacts even 40 dB below the music, yet some people are convinced they can hear artifacts such as jitter 100 dB down or lower. Compare that to test gear that can measure reliably down to the noise floor, and gives identical results every time.
24:43 Another factor is that our ears are most sensitive to frequencies in the treble range around 2 to 4 KHz. So distortion that lies mostly in that range is more noticeable and more objectionable than artifacts at lower frequencies. Intermodulation distortion typically contains both low and high frequencies, depending on the frequencies present in the music. Some IM components are not related musically to the fundamental pitches, so IM distortion is usually more objectionable and dissonant sounding than harmonic distortion. In a moment I'll play a DAW project that lets us hear the relative audibility of artifacts at different levels below music.
25:20 PROPER LISTENING TEST METHODS: With subjective listening tests, versus measuring, it's mandatory to change only the one thing being tested! For example, recording different performances to compare microphones or preamps is not valid because the performances can vary. The same subtle details we listen for when comparing gear, change from one performance to another. For example, a bell-like attack of a guitar note, or a certain sheen on a brushed cymbal. Nobody can play or sing exactly the same way twice, or remain perfectly stationary. So that's not a valid way to test preamps or anything else. Even if you could sing or play the same, a change in microphone position of even 1/4 inch is enough to make a real difference in the frequency response the mic captures.
26:03 Likewise, when the differences are subtle, non-blind (sighted) tests are invalid because, as JJ explained, we tend to hear what we want to hear, or think we should hear. This goes by many names—confirmation bias, placebo effect, buyer's remorse, and expectation bias. It's been said that audiophile reviewers can always identify which amplifier they're hearing—as long as they can read the name plate!
26:27 Likewise, if you record a rock band one day with preamp brand "A," and a jazz trio the next day with brand "B," it's impossible to assess anything meaningful about the preamps! One valid way to compare different preamps is to split the output of one microphone, and record each preamp to a separate track for comparative playback later. However, a splitter transformer can affect interaction between the microphone and preamp. So a better way is with re-amping, where you record the same playback through a loudspeaker of a single performance. Yes, the sound from the speaker may not be the same as a live cymbal or piano in the room. But that doesn't matter. The loudspeaker simply becomes the new "live" source, and any difference in tonality between the preamps being tested is still revealed.
27:09 Another self-testing method is called ABX. There are freeware software programs that play Wave files at random, which you try to identify. But you must repeat a test enough times to get a conclusive answer. If you happen to guess right one time that proves nothing—you'd have the same chance flipping a coin. Now, if you can get it right ten times out of ten, that's much more significant. One big feature of ABX testing is you can do it in the comfort of your own home, whenever you want. You can even test yourself over many months if you like. This avoids any chance of being "stressed" while you listen, which some people claim makes blind testing unreliable. Indeed, double-blind tests are the gold standard in every field of science. It amazes when some people claim that double-blind testing is not valid for assessing audio gear.
27:56 Besides changing only one thing at a time, matching the A and B volume levels is also important. When comparing two identical sources, the louder one often sounds better, unless it's already too loud. This is mostly due to the Fletcher-Munson effect bringing out more clarity and fullness at higher volume levels.
28:14 I can demonstrate a lot of things here in this video, but with lossy audio it may not be possible to hear the most subtle details. So I've explained how proper tests are conducted, and encourage you to try your own tests at home in your own familiar environment.
28:28 STACKING MYTH: People talk about "stacking" preamps and A/D/A converters in the sense that using the same preamp or converter for multiple tracks affects the result mix more than one preamp on one track does. Here, "stacking" means the preamps are used in parallel. Any coloration present in the preamp will be repeated for all of the tracks, so when all of the tracks are mixed together the result contains that coloration. So far so good—if the preamp used for every track has a 4 dB boost around 1 KHz, that's the same as using a flat preamp and adding an equalizer with 4 dB boost on the mix bus.
29:04 However, no competent preamp has a response nearly that skewed. Even modest gear is flat within 1 dB from 20 Hz to 20 KHz. But if a preamp does have a frequency response coloration—whether pleasing or not—it can be compensated for with mix bus EQ as just explained. It's not like mixing 20 tracks needs 20 times as much EQ to compensate!
29:25 Now let's consider distortion and noise—the other two audio parameters that affect the sound of a preamp or converter. Artifacts and other coloration from gear used in parallel does not add the same as when the devices are connected in series. In a little while you'll hear a mix that was sent through multiple A/D/A conversions in series to more easily hear the degradation. But this is not the same as stacking in parallel. In series is far more damaging.
29:48 This brings us to coherence. Noise and distortion on separate tracks do not add coherently. If you record the same mono guitar part on two analog tape tracks at once, when played back the signals combine to give 6 dB more output. But the tape noise is different on each track and so rises only 3 dB. This is the same as using a tape track that's twice as wide, or the difference between 8 tracks on 1/2 inch tape versus 8 tracks on 1 inch tape.
30:14 Likewise for distortion. The distortion added by a preamp or converter on a bass track has different content than the distortion added to a vocal track. So when you combine them in a mix, the relative distortion for each track remains the same. Thus, there is no "stacking" accumulation for distortion either. If you record a DI bass track through a preamp having 1 percent distortion on one track, then record a grand piano through the same preamp, the mixed result will have the same 1 percent distortion on each instrument.
30:45 PROPER TERMINOLOGY: Finally, subjective terms such as warm, cold, sterile, forward, and silky, and so forth, are not useful because they don't mean the same thing to everyone. Versus "3 dB down at 200 Hz" which is precise and leaves no room for interpretation. Of course, "warm" and "cold" or "sterile" could describe the relative amount of HF content. So saying "subdued or exaggerated highs" is better than "sterile" in my opinion.
31:13 Sometimes people refer to a piece of gear as being "musical" sounding or "resolving," but what does that really mean? What sounds musical to you may not sound musical to me. Some people like the added bass you get from an old-style receiver's Loudness switch. To me that usually sounds tubby, unless the music you're playing sounds thin. Same for a slight treble boost to add sheen, or a slight treble cut to reduce harshness. It's 1) highly dependent on the particular music playing, and 2) highly dependent on personal preference.
31:42 I don't think we need yet more adjectives to describe audio fidelity, when we already have perfectly good ones! The audiophile terms "PRaT" (Pace, Rhythm, and Timing) take this absurdity to new heights, because these words already have a specific musical meaning unrelated to whatever audiophiles think they are conveying.
32:00 PART 3 – Audio Examples
ARTIFACT AUDIBILITY: Now let's listen to some audio examples, so you can decide for yourself what matters.
[Flash www.ethanwiner.com/aes on screen] Again, all of the original audio files are on my web site. This first demo will help us determine at what level below the music distortion and noise can be heard.
32:18 SONAR: Artifact Audibility project file – For this first demo I created the nastiest noise I could muster, to serve as a worst case example of noise or distortion. If you can't hear this noise when it's 80 dB below the music, you won't hear other noises at that level such as jitter, digital aliasing, truncation distortion, or other types of distortion. I set the noise track's Volume to -20 because when played at full volume it is really irritating to hear! Have a listen. [Play the noise and slowly raise the Volume toward 0.]
32:44 Now, let's hear how audible this noise is when mixed in at various levels below music. [Play Men at Work with decreasing noise level.]
33:28 Okay, that was pop music already mastered and normalized to 1 dB below full scale. Now let's hear a soft gentle passage from my Cello Concerto, again adding the noise at various levels to tell when it can be heard. The average level of this soft passage is around -30, and peaks at only -15. Since the music is so soft at this point in the piece, the noise is actually 15 to 30 dB louder than its track Volume indicates.
34:32 DITHER: Now let's move on to dither. Does dither really have an audible effect, or is it just mental self-gratification? [Insert segment from live event, improvised, no script.]
36:58 Every time I've said in an audio forum that using dither has no audible affect on pop music recorded at sensible levels, it starts a huge fight. So let's listen and you can decide for yourself. To be clear, I don't advise against using dither (or recording at 24 bits)! We should always strive for the highest fidelity possible. And dither is free in all DAW software. But it's important to keep the value of things like dither in perspective. The distortion removed by dither is about 90 dB below full scale. So what I question is the claim that dither makes a "huge difference" in quality—especially as regards clarity and imaging and fullness—as some people report. I'd rather you decide based on a proper controlled test.
37:39 SONAR: Dither project file – To hear whether dither is audible when applied to typical pop music recorded at normal levels, I'll play two versions of a tune (Lullaby) I recorded and mixed in Cakewalk SONAR. One version was exported as 16 bits truncated, and the other was exported at 16 bits using SONAR's highest quality Pow-r 3 dither algorithm. I set up the Track Solo buttons in SONAR to toggle seamlessly between the two mixes. If anyone can reliably tell which track is dithered as I switch between these files, please raise your hand. [Then use Track Properties afterward to reveal the file names on screen.]
38:49 SONAR: Dither 2 project file – This next example is Handel Brockes from "Don M. of Novare." Converted in Sound Forge from 24 to 16 bits using either truncation or High-pass Triangular dither with High-pass contour noise shaping.
39:43 SONAR: Dither 3 project file – This next example is called God Speaks recorded by Lynn Fuston. Converted in Sound Forge from 24 to 16 bits using either truncation or High-pass Triangular dither with High-pass contour noise shaping. But the way, the arranger is J. Daniel Smith from Dallas.
40:45 When I want to test myself blind, I set up two parallel tracks in SONAR as I did here, with the Solo switches grouped while in opposite states. This lets me switch from one to the other with no clicks or pops. I put the mouse cursor over either Solo button, close my eyes, then click a bunch of times at random without paying attention to how many times I clicked. This way, I don't know which starting version I'm hearing. Then I listen carefully to see if I can really tell which is which as I switch back and forth. When I open my eyes I can see which track is currently solo'd.
41:15 SOUNDCARD QUALITY: This next demo lets us assess the relative quality of sound cards. Do we really have to pay thousands of dollars per channel for "transparent" audio? Is a "prosumer" sound card good enough to make recordings that sound fully professional?
41:28 Even "prosumer" sound cards these days are reasonably transparent, especially when compared to analog tape whose degradation anyone can hear after only one generation. Again, I'm not really suggesting that people sell off their high-end converters and replace them with $25 SoundBlaster cards! I have a SoundBlaster only because I need it to edit SoundFonts, which I still use for instrument samples. So it's not the sound card I normally use. But I believe it's a myth that inexpensive sound cards all sound terrible, as is commonly reported.
41:56 SONAR: \Sound Card Quality\Sound Cards file, Demo #1 – This first demonstration switches between two versions of the same performance I recorded in my fairly large home studio. This is a very old photo, but it shows the 34 by 18 foot size of my workspace. My friend Grekim Jennings is playing the acoustic guitar. The signal from my DPA 4090 microphone was split after the mic preamp. So the single microphone was recorded through a $25 SoundBlaster X-Fi sound card, and also through Grekim's Apogee 8000 at the same time.
42:47 SONAR: \Sound Card Quality\Sound Cards file, Demo #2 – This next demonstration compares a jazz recording as extracted from CD, versus a copy after one play/record generation through my M-Audio Delta 66 sound card. The recording is the Lynn Ariel trio, recorded by Tom Jung for a Yamaha demo CD when the Yamaha Promix 01R mixer was introduced. Even if you can hear a difference, is the quality really that terrible after one generation?
43:34 SONAR: \Sound Card Quality\SoundBlaster file – For this next demo I started with a rendered mix of one of my pop tunes (Men at Work), and recorded it out and back in again multiple times through the same SoundBlaster X-Fi card. We'll listen to the original mix, then after 1 play/record pass, 5 passes, 10 passes, and 20 passes. This way we can better hear how the sound is degraded.
45:06 Even 16 bits with no dither through a SoundBlaster beats the finest analog tape recorder in every way one could possibly assess audio fidelity. Arny Kruger once estimated in the rec.audio.pro newsgroup that analog tape has about the same resolution as 13 bits of digital.
45:22 I often see forum posts where someone is considering upgrading his sound card because he fears it's the weak link in his system. I always suggest this simple test: Take a CD that you think sounds amazing, and record it from your CD player's analog output into your sound card. Then play back the recording and see how it sounds compared to the original CD. If the recording sounds the same, that proves the sound card is transparent enough not to be the weak link.
45:48 BIT REDUCTION DEMO: For a final delivery medium, how many bits do we really need?
45:54 SONAR: Bit Reduction project file – For this demo I extracted a copy of Chic Corea's Spain recorded by the 12 cellists of the Berlin Philharmonic. If any type of music suffers from using too few bits, this track will reveal it very well. I'm using a freeware bit-reduction plug-in that can truncate audio in one-bit steps. As I slowly reduce the number of bits, see when you can first detect the first signs of degradation.
46:48 What type of degradation do you notice most as the quality is reduced below 16 bits? I hear mainly the addition of noise, and some gritty trebly distortion. But the overall tonality seem mostly unchanged until we get down to 8 or 9 bits.
47:02 I came across the freeware +decimate plug-in (www.soundhack.com/freeware.php) a few months ago when a forum fellow named Duggie insisted the benefit from recording at 24 bits becomes more obvious when multiple tracks are summed. I knew that stacking accumulation is a myth, so I found this "bit reducer" plug-in. Duggie was gracious enough to make two renders of his current tune—one with all tracks left at 24 bits, and another with this plug-in on every track set to 16 bits. Using this plug-in is equivalent to having recorded at 16 bits originally. Not only did the two mix files sound the same, and null with only minor artifacts way down in the noise floor, Duggie actually changed his mind and ended up agreeing with me. Now that's a rare occurrence for an audio forum!
47:45 PHASE SHIFT DEMO: Many people will tell you that phase shift is an audible problem. Some guy once bragged in a web forum that he can hear the phase shift in a ten-foot guitar cable.
[Insert segment from live event, improvised, no script.]
48:36 In my experience, usual amounts of phase shift are inaudible and benign. By usual amounts, I mean the phase shift that occurs normally in audio gear like preamps and power amps. This next demonstration will let you decide for yourself.
48:50 SONAR: Phase Shift project file, Demo #1 – To demonstrate the audibility of phase shift I'll use the freeware Sanford Phaser plug-in. This plug-in is functionally similar to the LittleLabs In-Between Phase hardware box you may know of, though it's more flexible. Most phase shifters vary the phase, using a single knob to control the center frequency. This type of device is also called an all-pass filter because it passes all frequencies equally, rather than boost or cut some frequencies as an equalizer does. The Sanford Phaser lets you choose how much phase shift to use, from 4 through 16 stages. For these tests I'll use 14 stages, which is a lot of phase shift! Most phase shifter effects boxes use 6 stages, and most amplifier circuits have less than one-tenth that amount.
50:03 For this demo we'll hear a recording of solo cello played by my friend Kate Dillingham. Listen as I enable the phase shifter and vary the left channel's turnover frequency. Note how the phase shift is audible only while the amount of shift is being varied. The timbre, or frequency response, is not changed. And as long as the shift is the same for both channels, imaging will not change either.
51:05 SONAR: Phase Shift project file, Demo #2 – For this next demo I'll use a funny recording made in 1962 by Dean Elliot called Lonesome Road. This is much more complex than a solo cello, and contains the full range of frequencies.
51:47 Earlier I mentioned phase shift effects boxes, and it's important to understand the difference between phase shift as demonstrated here, versus a "phase shifter" effect. A phase shifter effect is the same as this plug-in, but it combines the shifted output with the original input. That creates a comb filter frequency response having many peaks and nulls as we saw earlier. When you use a phase shifter effect, what you are hearing is the grossly affected frequency response, not the phase shift itself!
52:16 SONAR: Phase Shift project file, Demo #3 – The Sanford Phase Shifter has a Wet/Dry control to mix in the original signal. But first I'll set the music to mono, so I can adjust only one channel with the other channel muted. [Enable the GPan plug-in to play the left channel only in mono centered, and set the Phase Shifter plug-in Wet/Dry Mix to 0 (slider centered).] Now with the input and its shifted output combined, you can hear the familiar comb filter effect that changes the frequency response.
53:03 Understand that phase shift is not the same as a simple polarity reversal! Phase shift is the basis of all "normal" equalizers. By normal I mean minimum phase, not linear phase. Phase shift is a necessary and benign part of every analog equalizer's design.
53:19 The difference between phaser and flanger effects is the number of peaks and nulls. A phaser uses some number of all-pass filters in series, and the number of peaks and nulls is half the number of all-pass stages. A flanger effect instead uses a variable time delay, so the number of peaks and nulls is essentially infinite.
53:39 THE NULL TEST: A null test is absolute because it can identify all changes to an audio file, or the differences between two files. A null test can even identify distortion or other artifacts that we may not yet know to look for, as some people claim.
53:53 If two files can be nulled, then they are audibly identical—period, end of story. A null test can also be used to assess changes when using different signal wires, or adding a power conditioner, or anything else where you can record the same source material more than once. However, if files are recorded at different times they can drift apart over time and not remain nulled.
54:13 Null Test #1 project file – Some people believe that DAW summing is flawed because the quality is dependent on signal levels. I'll use this first example of a null test to disprove that myth. Modern DAW software uses 32-bit floating point calculations, which is very high resolution. As a result, the distortion is extremely low over a very wide range of signal levels. Some people claim that high levels in a DAW causes distortion at intermediate stages, as happens with analog gear. I can only speak for the SONAR software I use, but I never have this problem.
54:44 This example plays a mixed pop tune (Happy Go Lucky) through an EQ with 1 band cut. Then a second track with the same mix and EQ boosts the volume sent into the EQ by 18 dB. Then I patched in a freeware volume control after the EQ's output, lowering the volume by 18 dB to compensate. You'll hear that when summed together, the two versions null completely.
55:06 This test busts two myths: 1) That all plug-ins have a "sweet spot" where they sound best, and 2) that digital "summing" is somehow flawed. In my opinion, the only "sweet spot" that exists is due to the way Fletcher-Munson affects our hearing.
55:43 Null Test #2 project file – This next demo busts the myth that EQ cannot be countered exactly. For this test I put the same music on two tracks. One track has two equalizers in series, but with opposite settings. That is, one has three bands of cut, and the other has those same bands set to boost. Even after going through both EQ plug-ins, the result nulls perfectly with an unmodified version of the same mix.
56:28 Again, small timing errors, drift, and phase shift can prevent two otherwise identical-sounding files from nulling, even though the drift and phase shift themselves are inaudible. Drift won't happen in DAW software, but it does happen with analog tape and vinyl records. You can also get small amounts of drift recording the same thing several times digitally, due to minor clock variations that are otherwise inaudible. I'll load the earlier file that recorded an acoustic guitar through a SoundBlaster and Apogee converter at the same time to show this. These were recorded into different sound cards, which means they had different clock sources.
57:03 Null Test #3 – SONAR: \Sound Card Quality\Sound Cards file, Demo #1 – For this demo I'll turn off the Solo on both tracks, then reverse the polarity of Track 2. Now you can hear the two recordings drift in and out of the null state.
57:30 The EQ reversal null tests also tells us that simple response deviations, such as a microphone's low frequency proximity boost, can be countered with EQ. It's much more difficult to counter acoustic issues caused by reflections, because comb filtering is much more complex. But simple one- or two-pole curves can be undone with EQ. In tech-speak, a "pole" is simply one stage having 6 dB per octave.
57:56 Many microphones include a high frequency "presence" boost around 5 KHz. There's no real difference between that and a flat microphone with EQ added, unless the boost is caused by the capsule ringing. When my partner Doug Ferrara bought a Neumann TLM103, I brought my audiotechnica 4033 to his studio. We set the microphones side by side and I sang into both at once. After we applied an EQ curve to the 4033 matching the boost shown in the TLM manual, both microphones then sounded more or less the same."