A great podcast / radio documentary on how sports events are mixed (and sound effects are added) to make the experience on your television “better than being there”.
Category: recordings
A listening primer on reverb
What’s interesting is to see the old photos of Andy Moorer…
What’s even more interesting is to know that they named it “Deep note” after “Deep thought” from HHGG… Small things make me happy.
B&O Tech: What is “Beam Width Control”?
#34 in a series of articles about the technology behind Bang & Olufsen loudspeakers
A little background:
Distance Perception in “Real Life”
Go to the middle of a snow-covered frozen lake with a loudspeaker, a chair, and a friend. Sit on the chair, close your eyes and get your friend to place the loudspeaker some distance from you. Keep your eyes closed, play some sounds out of the loudspeaker and try to estimate how far away it is. You will be wrong (unless you’re VERY lucky). Why? It’s because, in real life with real sources in real spaces, distance information (in other words, the information that tells you how far away a sound source is) comes mainly from the relationship between the direct sound and the early reflections that come at you horizontally. If you get the direct sound only, then you get no distance information. Add the early reflections and you can very easily tell how far away it is. If you’re interested in digging into this on a more geeky level, this report is a good starting point.
A little more background:
Distance perception in a recording
Recording engineers use this information as a trick to simulate differences in apparent distance to sound sources in a stereo recording by playing with the so-called “dry-wet” ratio – in other words, the relative levels of the direct sound and the reverb. I first learned this in the little booklet that came with my first piece of recording gear – an Alesis Microverb (1st generation… It was a while ago…). To be honest – this is a bit of an over-simplification, but it’s good enough to work (for example, listen to the reverberation on Grover’s voice change as he moves from “near” to “far” in this video). The people at another reverb unit manufacturer know that the truth requires a little more details. For example, their flagship reverb unit uses correctly-positioned and correctly-delayed early reflections to deliver a believable room size and sound source location in that room.
Recording Studios vs. Living Rooms
When a recording engineer makes a recording in a well-designed studio, he or she is sitting not only in a carefully-designed acoustical space, but a very special area within that space. In many recording studios, there is an area behind the mixing console where there are no (or at least almost no) reflections from the sidewalls . This is accomplished either by putting acoustically absorptive materials on the walls to soak up the sound so it cannot reflect (as shown in Figure 1), or to angle the walls so that the reflections are directed away from the listening position (as shown in Figure 2).
Both of these are significantly different from what happens in a typical domestic listening room (in other words, your living room) where the walls on either side of the listening position are usually acoustically reflective, as is shown in Figure 3.
In order to get the same acoustical behaviour at the listening position in your living room that the recording engineer had in the studio, we will have to reduce the amount of energy that is reflected off the side walls. If we do not want to change the room, one way to do this is to change the behaviour of the loudspeaker by focusing the beam of sound so that it stays directed at the listening position, but it sends less sound to the sides, towards the walls, as is shown in Figure 4.
So, if you could reduce the width of the beam of sound directed out the front of the loudspeaker to be narrower to reduce the level of sidewall reflections, you would get a more accurate representation of the sound the recording engineer heard when the recording was made. This is because, although you still have sidewalls that are reflective, there is less energy going towards them that will reflect to the listening position.
However, if you’re sharing your music with friends or family, depending on where people are sitting, the beam may be too narrow to ensure that everyone has the same experience. In this case, it may be desirable to make the loudspeaker’s sound beam wider. Of course, this can be extended to its extreme where the loudspeaker’s beam width is extended to radiate sound in all directions equally. This may be a good setting for cases where you have many people moving around the listening space, as may be the case at a party, for example.
For the past 5 or 6 years, we in the acoustics department at Bang & Olufsen have been working on a loudspeaker technology that allows us to change this radiation pattern using a system we call Beam Width Control. Using lots of DSP power, racks of amplifiers, and loudspeaker drivers, we are able to not only create the beam width that we want (or switch on-the-fly between different beam widths), but we can do so over a wide frequency range. This allows us to listen to the results, and design the directivity pattern of a loudspeaker, just as we currently design its timbral characteristics by sculpting its magnitude response. This means that we can not only decide how a loudspeaker “sounds” – but how it represents the spatial properties of the recording.
What does Beam Width Control do?
Let’s start by taking a simple recording – Susanne Vega singing “Tom’s Diner”. This is a song that consists only of a fairly dryly-recorded voice without any accompanying instruments. If you play this tune over “normal” multi-way loudspeakers, the distance to the voice can (depending on the specifics of the loudspeakers and the listening room’s reflective surfaces) sound a little odd. As I discussed in more detail in this article, different beam widths (or, if you’re a little geeky – “differences in directivity”) at different frequency bands can cause artefacts like Vega’s “t’s” and “‘s’s” appearing to be closer to you than her vowel sounds, as I have tried to represent in Figure 5.
If you then switch to a loudspeaker with a narrow beam width (such as that shown in the directivity plot in Figure 7 – the beam width is the vertical thickness of the shape in the plot – note that it’s wide in the low frequencies and narrowest at 10,000 Hz), you don’t get much energy reflected off the side walls of the listening room. You should also notice that the contour lines are almost parallel, which means that the same beam width doesn’t change as much with frequency.
Since there is very little reflected energy in the recording itself, the result is that the voice seems to float in space as a pinpoint, roughly half-way between the listening position and the loudspeakers – much as was the case of the sound of your friend on the snow-covered lake. In addition, as you can see in Figure 7, the beam width of the loudspeaker’s radiation is almost the same at all frequencies – which means that, not only does Vega’s voice float in a location between you and the loudspeakers, but all frequency bands of her voice appear to be the same distance from you. This is represented in Figure 8.
If we then switch to a completely different beam width that sends sound in all directions, making a kind of omnidirectional loudspeaker (with a directivity response as is shown in Figure 9), then there are at least three significant changes in the perceived sound. (If you’re familiar with such plots, you’ll be able to see the “lobing” and diffraction caused by various things, including the hard corners on our MDF loudspeaker enclosures. See this article for more information about this little issue… )
The first big change is that the timbre of the voice is considerably different – particularly in the mid-range (although you could easily argue that this particular recording only has mid-range…). This is caused by the “addition” of reflections from the listening room’s walls at the listening position (since we’re now sending more energy towards the room boundaries). The second change is in the apparent distance to the voice. It now appears to be floating at a distance that is the same as the distance to the loudspeakers from the listening position. (In other words, she moved away from you…). The third change is in the apparent width of the phantom image – it becomes much wider and “fuzzier” – like a slightly wide cloud floating between the loudspeakers (instead of a pin-point location). The total result is represented in Figure 10, below.
All three of these artefacts are the result of the increased energy from the wall reflections.
Of course, we don’t need to go from a very narrow to an omnidirectional beam width. We could find a “middle ground” – similar to the 180º beam width of BeoLab 5 and call that “wide”. The result of this is shown in Figures 11 and 12, with a measurement of the BeoLab 5’s directivity shown for comparison in Figure 13.
If we do the same comparison using a more complex mix (say, Jennifer Warnes singing “Bird on a Wire” for example) the difference in the spatial representation is something like that which is shown in Figures 14 and 15. (Compare these to the map shown in this article.) Please note that these are merely an “artist’s rendition” of the effect and should not be taken as precise representations of the perceived spatial representation of the mixes. Actual results will certainly vary from listener to listener, room to room, and with changes in loudspeaker placement relative to room boundaries.
Of course, everything I’ve said above assumes that you’re sitting in the “sweet spot” – a location equidistant to the two loudspeakers at which both loudspeakers are aimed. If you’re not, then the perceived differences between the “narrow” and “omni” beam widths will be very different… This is because you’re sitting outside the narrow beam, so, for starters, the direct sound from the loudspeakers in omni mode will be louder than when they’re in narrow mode. In an extreme case, if you’re in “narrow” mode, with the loudspeaker pointing at the wall instead of the listening position, then the reflection will be louder than the direct sound – but now I’m getting pedantic.
Wrapping up…
The idea here is that we’re experimenting on building a loudspeaker that can deliver a narrow beam width so that, if you’re like me – the kind of person who has one chair and no friends, and you know what a “stereo sweet spot” is, then you can sit in that chair and hear the same spatial representation that the recording engineer heard in the recording studio (without having to make changes to your living room’s acoustical treatment). However, if you do happen to have some friends visiting, you have the option of switching over to a wider beam width so that everyone shares a more similar experience. It won’t sound as good (whatever that might mean to you…) in the sweet spot, but it might sound better if you’re somewhere else. Similarly, if you take that to an extreme and have a LOT of friends over, you can use the “omni” beam width and get a more even distribution of background music throughout the room.
For more information on Beam Width Control
Shark Fins and the birth of Beam Width Control
Post-script
For an outsider’s view, please see the following…
“Ny lydteknikk fra Bang & Olufsen” – Lyd & Bilde (Norway)
Stereophile magazine (October 2015, Print edition) did an article on their experiences hearing the prototypes as well.
“BeoLab 90: B&O laver banebrydende højttaler” – Lyd & Bilde (Norway)
Slow-motion Vinyl
Voices in the Wire
A GREAT podcast from 99% Invisible about Tony Schwartz’s recording history and the world’s first “commercial” radio station that started in Frank Conrad’s Garage.
CHECK YOUR PREFERENCES! (was: “Garbage in, garbage out…”)
Stop the Press
The following posting is a description of a test that I did on two streaming services where I concluded that one of them is probably doing dynamic range compression and the other one probably isn’t.
My conclusion was only partly correct. One of the players was, indeed, doing dynamic range compression. However, this was because I asked it to do so. Since posting this article, I have found an “Advanced Preferences” in the player that includes the option to make the tracks more alike in level. I had this option set to “ON” when I did this test.
So, almost everything you are about to read below is technically correct – but the conclusion that one of the services is “worse” than the other is not fair. I’ll re-do the entire test and re-post the new analysis and conclusions…
My humble apologies….
-geoff
Introduction
Tom Lehrer once said “Life is like a sewer. What you get out of it depends on what you put into it.” The same is true when you’re testing loudspeakers. What you get out of them depends on what you feed into them.
This is an important thing to keep in mind when you’re listening to loudspeakers. If listen to a pair of loudspeakers for the first time and you decide that there’s something about them that you don’t like, you should make sure that whatever it is that you don’t like about them is not the fault of your source. This might mean that there’s something in the recording that’s causing the issue (like a resonance in the vocal booth, or a clipped master, for example). It might mean that the player has some issues (computer headphone outputs are notoriously noisy, for example). It might mean that you should be careful that you’re not listening to a lower bitrate psychoacoustic CODEC like MP3, for example. Or it might mean something else that isn’t immediately obvious.
This is particularly true if you use a streaming service, of which there seem to be newcomers every day… Now, before we go any further, I’ll warn you that I’m not going to mention any names in this posting. So, don’t email me and ask “Which services were you talking about in that posting? I promise I won’t tell anyone…” There are various reasons for this – just don’t bother asking.
When you choose a streaming service, there are a couple of things to consider. One is the level of service (and therefore, possibly quality) you want to buy. Some services offer better quality for a higher price. Some services offer music streaming for free if you can live with the commercials (typically at the same audio quality as the low-price version). Different services use different CODEC’s. Some use MP3, some use Ogg Vorbis, and some claim to stream lossless signals. You can then choose different bitrates or CODEC’s at different prices.
I’m not going to get into a discussion between the pro’s and con’s of 320 kbps MP3 vs 160 kbps Ogg Vorbis, or whether anything is as “good” as lossless. Lots of people are discussing that on lots of websites. Feel free to go there to have that fight. What I want to talk about is something else: dynamic range compression.
You should beware when you choose a streaming service that some of them apply a dynamic range compression to the music, regardless of the CODEC. This not only means that the quiet tunes are louder, and the louder tunes are quieter. It also means that the quieter sections of the tunes are louder, and the louder sections are quieter. Even worse, in some cases, this my be done on a frequency-dependent basis (using something called a multi-band compressor) which changes the overall spectral balance of the tune over time, and does it differently for different tracks due to their different spectral characteristics.
Since I have no plan to point fingers, my goal here is to help you learn to point your own fingers. In other words, how can you easily find out whether your streaming service is doing more than “just” reducing the bitrate of the audio signal.
The trick to find this out is a simple matter of doing a quick back-to back comparison. I’ll describe the technique assuming that you’re comparing two streaming services using a single computer – but you could (of course) apply this technique to a streaming service vs. local file playback.
- Open up accounts on two streaming services.
- Simultaneously open a player for each service.
- Set the volume in each player to maximum.
- Set the output volume of your computer (or the master volume of the loudspeakers/headphones) to a comfortable level. (We’ll have to assume that your computer isn’t doing anything with dynamic range, which it probably is…)
- Open up the same song on both players (a suggested list is below)
- Go back and forth, playing 10 (or so) seconds on one player, stopping, then playing the same 10 seconds on the other player.
- Listen to what happens when you play the same section of the same song in the two players. (Take notes, if necessary)
- Load a very different song and repeat steps 6. and 7.
For example, open up Player “A” and Player “B” and load “Arlandria” by the Foo Fighters.
Let’s say that, when you go back and forth, the song is quieter in Player A than in Player B (see Figure 1, below, as an example…). This might be for different reasons (like the volume knobs are not aligned, for example…)
Now load “Bird on a Wire” by Jennifer Warnes and play that (which is mastered to have a much lower average (RMS) level than Arlandria – therefore it should be quieter…).
Let’s say that, when you go back and forth, the song is now louder in Player A than in Player B (as is shown in Figure 2). This means that the “volume knob” theory above doesn’t hold (if it did, Player A would always be quieter than Player B).
So, Arlandria is louder in “B” but Bird on a Wire is louder in “A”. This means that at least one (perhaps both) of the two players is doing something with the dynamic range of the tunes. What is most likely is that at least one of the players is making the Foo Fighters too quiet and Jennifer Warnes too loud (relative to their original mastering).
There might be a good reason for this. It could be that Player “A” is designed to be a radio station – good for background music where you press “play” and you never have to chase the signal with the volume control (in other words, you don’t have to suddenly drop the volume during dinner because the Foo Fighters started playing). However, there might be side effects, since it is likely that the dynamic range compression in Player A (assuming for simplicity’s sake that it is only Player A in this example that is mucking around with the levels) is happening on the signal in real-time – and is not based on a gain offset on a per-tune basis (as some software players do…)
So, as you can see, there’s not much of a trick here. It’s just a matter of doing a direct player-to-player comparison, making sure that as many components in the signal path remain constant. (For example, be careful that your computer doesn’t have a compressor on its headphone output if that’s what you’re using. Many computer headphone outputs have some kind of dynamic range stuff happening like noise gating, peak compression / soft clipping or just general envelope shaping.) Best to use a digital output from your computer to a reliable DAC that just does what it’s told, and doesn’t add any fancy effects like volume levelling.
Some analysis
I took two streaming services (each of them delivering their “high quality” option) and routed their outputs to Max/MSP using Soundflower and recorded each as a 16-bit .wav file. Both were set to maximum volume, as was the master volume of my MacBook Pro.
I then used Audacity to manually time-align and trim the files to have exactly the same length (in samples). The length of each file was arbitrary and based more on when the chorus ended than anything else…
For the analyses of peak level (listed in dB FS, even though I know that dB FS is only supposed to be used for RMS measurements… sue me…), total RMS level, and crest factor, I just wrote a small routine in Matlab to look after this for me. The two values shown are for the Left and Right channels for each tune.
I’ve also supplied a screen shot of the entire tracks analysed, shown in Audacity, so you can get an intuitive feel of the difference between the two streaming services.
Some things to look out for here: If the difference in the peak amplitudes matches the difference in the RMS values (and therefore the crest factors match each other) then the differences in the output levels of the two streaming services is the result of a simple change in level for the entire tune – as if someone had changed the volume setting and left it alone for the whole song (this is the technique ostensibly used by at least one ubiquitous computer-based audio player). However, if the crest factors are different for the two streaming services for a given song (meaning that the differences in the peak amplitudes and the RMS values are different from service to service for a given song) then this means that some time-variant dynamic processing is going on – probably something like a compressor, a peak limiter, a soft-clipper, or just a clipper.
Foo Fighters – Arlandia (first 81.67 seconds)
[table]
Streaming Service “A”, Left, Right
Peak Amplitudes (dB FS), -4.2209, -4.1578
RMS (dB FS), -15.6585, -15.2328
Crest factor (dB), 11.4376, 11.0750
[/table]
[table]
Streaming Service “B”, Left, Right
Peak Amplitudes (dB FS), 0.0, 0.0
RMS (dB FS), -7.4978, -7.1929
Crest factor (dB), 7.4978, 7.1929
[/table]
Jennifer Warnes – Bird on a Wire (first 59.43 seconds)
[table]
Streaming Service “A”, Left, Right
Peak Amplitudes (dB FS), 0.0, 0.0
RMS (dB FS), -9.4162, -10.6752
Crest factor (dB), 9.4162, 10.6752
[/table]
[table]
Streaming Service “B”, Left, Right
Peak Amplitudes (dB FS), -0.1287, -0.3205
RMS (dB FS), -16.5152, -17.8888
Crest factor (dB), 16.3865, 17.5683
[/table]
Lorde – Royals (first 73.46 seconds)
[table]
Streaming Service “A”, Left, Right
Peak Amplitudes (dB FS), -0.0863, -0.0863
RMS (dB FS), -11.0863, -11.0712
Crest factor (dB), 11.0000, 10.9849
[/table]
[table]
Streaming Service “B”, Left, Right
Peak Amplitudes (dB FS), -0.0005, -0.0005
RMS (dB FS), -11.0509, -11.0365
Crest factor (dB), 11.0504, 11.0360
[/table]
So, what confuses me at the end of this is that Royals is almost identical in both. And, before you jump to conclusions, I checked… I did not screw up the captures. When I subtract “B” from “A”, I can hear artefacts typical of a psychoacoustic CODEC, so I know that the two are different enough to be measurably different. What doesn’t make sense is that, if “A” is dynamically compressing the other two tracks more than “B” (which it appears to be), then why is it not messing around with “Royals”? Hmmmm… I guess that more investigation is necessary with respect to this little anomaly – but not today… Sorry.
We can also check to see if the dynamic range compression that’s being used is a multi-band type or not by doing a quick spectral analysis of the excerpts. I took each of the 6 files (three songs x 2 streaming services) and did the following:
- Filtered the left channel of each track using 3rd-order butterworth highpass and lowpass filters with cutoff frequencies between the ISO octave-band centres ranging from 31.5 Hz to 16 kHz inclusive.
- Calculated the total RMS value of each of the entire band-limited versions of the tracks
- Plotted the results in dB FS
Now, before you start complaining about my methodology here, remember that I’m just using a “big paint brush” approach to checking this out. I’m just interested in seeing if there are any significant differences between the spectral balances of a single track coming from the two streaming services. The results are plotted below.
What you can see in those three plots is that the spectra of a given tune from the two services are roughly the same – only the overall levels are different (which roughly correspond to the RMS values shown in Figure 4).
In case you’re curious, if you level-align the plots in Figures 5 through 7 inclusive, you’ll see that the worst-case differences in spectra for the three tunes across the entire frequency range analysed are about 0.7 dB for Arlandria, 0.9 dB for Bird on a Wire, and 0.08 dB for Royals. However, be a little careful here, since this is an RMS value of the entire track – and may not reflect the more “instantaneous” spectral differences, say, on a clipped peak in “Bird on a Wire”, for example. I won’t bother to analyse those, since it would not be surprising for a peak compressor to generate distortion-like harmonics when it’s changing its gain. This could result in a spectral difference that looks like a multi-band compressor, but isn’t. Windowing can be distracting sometimes…
The conclusion
One moral of the story here is that knowing the psychoacoustic CODEC and bitrate used by your streaming service is not necessarily enough information to know whether or not you’re getting the audio quality you may want.
The other moral is that you shouldn’t necessarily trust that a “high quality” version of a streaming service is the same as the “high quality” version of another streaming service.
Another message here is that you should not compare one service to another one and think that you’re only comparing CODEC’s – for example, don’t listen to Deezer vs. Spotify vs. Tidal in order to find out if you can hear the difference between MP3, Ogg Vorbis, and lossless. You might hear a difference, but it might not be caused by the CODEC’s…
Finally, an important moral is: if you’re using a streaming service to test loudspeakers (for example, in a store) – make sure that the version of the tune that you’re listening to is the version that you know – from the same streaming service and the same quality level. Otherwise, the problems (or just differences) you might hear might not be the fault of the speaker… (For example, if your streaming service at home is “worse” than the store’s – it might make the speakers sound better then you think just because the source is better…) Remember garbage in = garbage out, regardless of whether the description is more applicable to the loudspeakers at the store or the ones at home.
Post-script: Comparison to CD
I thought it might be interesting to compare a CD version to the streaming versions. Unfortunately, the only disc I have of the three tracks here is the Foo Fighters. I have the original Famous Blue Raincoat album from once-upon-a-time, but I don’t have the 20th anniversary re-release, which is what the streaming services appear to be using.
[table]
Streaming Service “A”, Left, Right
Peak Amplitudes (dB FS), -4.2209, -4.1578
RMS (dB FS), -15.6585, -15.2328
Crest factor (dB), 11.4376, 11.0750
[/table]
[table]
Streaming Service “B”, Left, Right
Peak Amplitudes (dB FS), 0.0, 0.0
RMS (dB FS), -7.4978, -7.1929
Crest factor (dB), 7.4978, 7.1929
[/table]
[table]
Ripped from CD, Left, Right
Peak Amplitudes (dB FS), -0.0003, -0.0003
RMS (dB FS), -10.7888, -10.3700
Crest factor (dB), 10.7885, 10.3697
[/table]
So, the data shows that, at least for this tune, the version ripped from a CD is different again from the two streaming services. This can also be seen if we put the three versions side-by-side on a DAW.
As can be seen below, this means that some samples are clipped in the process in the streaming service that are not clipped in the original…
An interesting listening test…
What’s really interesting is to repeat the test on different equipment such as headphones and loudspeakers, or different DAC’s – or even different listening levels.
(The questions are randomised each time you re-load the page – but of course there’s a learning effect that might influence your results)
Le Studio decays
Speaking of Vinyl…
