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More => Old Versions => Media Center 15 (Development Ended) => Topic started by: mojave on July 16, 2010, 11:23:43 am
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My hard drive crashed recently and I had to start from scratch. This was a work computer so personal stuff, like MC and music, was not backed up. I'm finally getting around to testing the latest Bass Management crossovers.
I downloaded and installed the Voxengo Span VST plugin to check the frequency response and levels. I also downloaded this Pink Noise (http://www.ethanwiner.com/misc-content/pinknoise.wav) file from Ethan Winer's website. I set it to repeat in MC and then check things out in Span while playing with Room Correction. The default settings in Span have a tilted frequency response. To get a flat response one needs to click the Edit button in the top right corner of the plugin and change the slope to 2.90. I change the Freq Low to 60 and the Freq High to 2.00K so the window is narrower. Also, the pink noise starts to roll off at 60 Hz. I then change the Type (still in Edit in Span) to Avg so I get a smoother line in the spectrum. I do this for both the L/R group and the LFE group.
First I played the pink noise with the L & R channels combined and Room Correction off to make sure I had a flat frequency response. Then I enabled Room Correction with the Bass Management set to the following: Crossover of 200 Hz (to better see what is going on), Move, 24 dB/octave for both High and Low Pass filters. The bass is rolling off and is not as loud.
I then kept increasing the Subwoofer channel in Room Correction until I was back to a flat frequency response. I had to increase the gain by 6 dB in order to have a flat frequency response. Remember, this is the electronic signal and has nothing to do with the speakers, subwoofers, and room. You are trying to reproduce the original signal, but with it split to the the mains and the subwoofer.
It may sum correctly with a multichannel source, but it isn't summing correctly with a stereo source. I'll check the multichannel stuff next.
I also noticed that all crossover settings were lower in frequency than what was set. When the Subwoofer channel had the gain increased, then the crossover was closer to its actual setting.
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Here are some more charts showing the crossover.
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Here is the final chart with L, R, LFE combined and the Subwoofer with a 6 dB gain.
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I then kept increasing the Subwoofer channel in Room Correction until I was back to a flat frequency response. I had to increase the gain by 6 dB in order to have a flat frequency response. Remember, this is the electronic signal and has nothing to do with the speakers, subwoofers, and room. You are trying to reproduce the original signal, but with it split to the the mains and the subwoofer.
Remember that the moved bass gets attenuated by 10 dB. This is done because the LFE channel should be 10 dB louder at equivalent signal levels.
But since there's bass coming from the left and right, together they're shifted roughly +6 dB.
The math looks like:
LFE = ([Left bass] - 10 dB) + ([Right bass] - 10 dB)
So the results you're posting seem reasonable to me. Could you explain in more detail if you disagree?
Thanks.
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I also noticed that all crossover settings were lower in frequency than what was set. When the Subwoofer channel had the gain increased, then the crossover was closer to its actual setting.
Here's a quote from AlexB that might explain this:
This is probably obvious and you may already have it correct, but at the crossover frequency the curves are supposed to cross at -6 dB.
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Also, thank you for all your testing and help with this feature. I really appreciate it.
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Remember that the moved bass gets attenuated by 10 dB. This is done because the LFE channel should be 10 dB louder at equivalent signal levels.
But since there's bass coming from the left and right, together they're shifted roughly +6 dB.
The math looks like:
LFE = ([Left bass] - 10 dB) + ([Right bass] - 10 dB)
So the results you're posting seem reasonable to me. Could you explain in more detail if you disagree?
Thanks.
Let's solve the math.
LFE = ([Left bass] - 10 dB) + ([Right bass] - 10 dB)
LFE = ([Left bass @ 75 db] - 10 dB) + ([Right bass @ 75 db] - 10 dB)
LFE = (75 dB-10 dB)+(75 dB-10 dB)
LFE = (74.99 dB) + (74.99 dB)
LFE = 81.015714067 dB
L @ 75 + R @ 75 also equals 81 dB so the bass matches the mains (at least on paper). Therefore, I agree completely with what you are saying. When I electronically take the L + R + LFE channel, though, the bass is 6 dB too low as can be seen in the chart in the first post.
I couldn't find another multichannel spectrum analyzer. I double checked my settings and ran everything again. The bass is still 6 dB too low. This is confirmed two ways. First, the crossover is at the wrong frequency until the the bass is increased. Second, the combined channels show a drop off in volume well below the crossover region.
Here's a quote from AlexB that might explain this:
Yes, the curves do cross at around -6 dB as can be seen in my charts. The problem is that the bass does not come back up to the correct volume.
Again, the goal is to have the original signal look exactly the same after the stereo channel has been split to the LFE channels and then all are combined again in the spectrum analyzer. This gives an indication of what it will sound like with a properly calibrated system (subwoofer and speakers set to the same volume level).
Either I'm measuring wrong or you are summing wrong.
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Let's solve the math.
LFE = ([Left bass] - 10 dB) + ([Right bass] - 10 dB)
LFE = ([Left bass @ 75 db] - 10 dB) + ([Right bass @ 75 db] - 10 dB)
LFE = (75 dB-10 dB)+(75 dB-10 dB)
LFE = (74.99 dB) + (74.99 dB)
LFE = 81.015714067 dB
L @ 75 + R @ 75 also equals 81 dB so the bass matches the mains (at least on paper). Therefore, I agree completely with what you are saying. When I electronically take the L + R + LFE channel, though, the bass is 6 dB too low as can be seen in the chart in the first post.
I got a little lost there.
Shouldn't it be:
75 dB - 10 dB = 65 dB
65 dB + 65 dB = 71 dB
It almost seems like we should average the redirected bass instead of sum it, so that it would always be -10 dB instead of, in this case, -4 dB.
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Just like 65 plus 65 equals 71 dB instead of 130, you can't just subtract 10 dB from 75 to get 65 dB. You have to convert to Pascals and then add/subtract and then convert back to decibels. Here is one for doing the conversion (http://www.sengpielaudio.com/calculator-soundlevel.htm). There are others available, too.
Another thing to consider is that I am talking about 2 channel to 2.1. When you combine with an already existing LFE channel the level mixing is different.
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When we attenuate the LFE by -10 dB, it means we multiply by about ~0.32 in the PCM space (which is analogous to voltage / pressure).
So with one volt:
LFE = ([Left bass] - 10 dB) + ([Right bass] - 10 dB)
LFE = (1.0 * 0.32) + (1.0 * 0.32)
LFE = 0.64
1.0 volt becoming 0.64 is about a -4 dB shift, which matches my previous post.
So I'm still not clear on the problem. Could you spell it out in a little more detail, maybe with math?
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you can't just subtract 10 dB from 75 to get 65 dB
Maybe -10 dB means something different in different contexts.
To me, it means the output signal is ten decibels less than the input signal. So you can "just subtract" as that's the point. Of course, you have to take care to do the math correctly to ensure a -10 dB shift.
Is it possible you're thinking about adding a signal that's 10 dB quieter or something? Is there some other angle to this?
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I think you would need to do
(0.707946 x L) + (0.707946 x R)
to get a flat frequency response curve when the bass from 2.0 is redirected to 0.1.
Before the redirection you had two output channels that added to each other. After the redirection you have only one output channel so it should receive a louder signal than one of the two outputs channels received before (the above math results ~1.41). [edited this part a bit]
Of course you can't go over 0 dBFS in the digital domain so you would need to attenuate all channels to compensate.
This is done because the LFE channel should be 10 dB louder at equivalent signal levels.
Matt, isn't this the Dolby standard for movie theaters? I don't think it is normally assumed that the subwoofer speaker is set to be 10 dB louder than the other speakers. For instance, when a 5.1 channel audio recording is mixed, I think the assumption is that the speakers are configured to produce a flat response curve.
I suggested the LFE mixer so that the balance between the redirected bass and the bass that comes directly from the source 0.1 channel could be freely adjusted.
Here's a quote from http://www.hometheaterhifi.com/technical-articles/163-the-misunderstood-01-lfe-channel-in-51-digital-surround-sound.html :
Stop! Before you run home and set your subwoofer 10 dB higher than you've already set it, you need to read on and find out why this is not necessary.
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Matt, isn't this the Dolby standard for movie theaters? I don't think it is normally assumed that the subwoofer speaker is always set to be 10 dB louder than the other speakers. For instance, when a 5.1 channel audio recording is mixed, I think the assumption is that the speakers are configured to produce a flat response curve.
I suggested the LFE mixer so that the balance between the redirected bass and the bass that comes directly from the source 0.1 channel could be freely adjusted.
Here's a quote from http://www.hometheaterhifi.com/technical-articles/163-the-misunderstood-01-lfe-channel-in-51-digital-surround-sound.html :
So does anyone know if AC3Filter or FFDShow already boost the LFE by 10 dB relative to the other channels in analog output mode?
It didn't seem to me like they would because then you either have to worry about clipping or have to attenuate all the other channels -10 dB.
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Of course you can't over 0 dBFS in the digital domain so you would need to attenuate all channels to compensate.
Isn't this another argument for the 10 dB shift on the subwoofer channel?
Otherwise, we're going to have to really crank down the volume on all the other channels if we're redirecting from 5.1 or 7.1 so the subwoofer can't clip.
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So does anyone know if AC3Filter or FFDShow already boost the LFE by 10 dB relative to the other channels in analog output mode?
It didn't seem to me like they would because then you either have to worry about clipping or have to attenuate all the other channels -10 dB.
Google search: +10dB LFE site:.ac3filter.net
Seems like it has been discussed there. I have not yet read the discussions.
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Otherwise, we're going to have to really crank down the volume on all the other channels if we're redirecting from 5.1 or 7.1 so the subwoofer can't clip.
I think you have to do that. Here's an example from: http://ac3filter.net/guides/mixing_matrix
Matrix normalization
Note that in the example above we mix several input channels into one output channel. But imagine a loud 'boom' sound reproduced by all channels. In this case when we sum several channels into one, overflow may occur.
Overflow means distortions (see 'Loudness and dynamic range' article for details). What to do if we don't want distortions? We should decrease the sound level. To do this we may decrease Master gain, or lower all values in the matrix. To do this automatically, use matrix normalization. See the result of normalization on 5.1 to stereo downmix matrix:
Not normalized:
(http://ac3filter.net/files/images/matrix-not-normalized.png)
Normalized:
(http://ac3filter.net/files/images/matrix-normalized.png)
Normalized matrix looks more complicated, but in fact it does exactly the same thing as unnormalized, just a little quieter. Normalization is enabled by default. For clearness we will use unnormalized matrix afterwards, but for real use, it is better to use normalized matrix.
Maximum level of an output channel equals to the sum of the row elements:
MaxLevel(channel) = Sum(Row(channel))
To avoid overflow we must make the level of all output channels <= 1. To do this we should divide all matrix elements to the maximum level among all channels:
MaxLevel = Max(MaxLevel(channel)) NormalizedMatrix[i][j] = 1/MaxLevel * Matrix[i][j]
That's exactly what matrix normalization option does.
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I think you have to do that. Here's an example from: http://ac3filter.net/guides/mixing_matrix
We use normalized mixing matrixes for JRSS downmixing, and I think you're right that we need to do the same here as well.
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Google search: +10dB LFE site:.ac3filter.net
Seems like it has been discussed there. I have not yet read the discussions.
From my reading, it sounds like it assumes either your amp or subwoofer will make the LFE 10 dB louder.
So that means when mixing from 2.0 to 2.1 with bass redirection, the LFE should be attenuated by 10 dB so it matches the levels you would get when playing a movie.
I suppose we could make additional options for "Move / Copy bass to subwoofer (LFE) at -0 dB".
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I just sketched out the updated math, and I think I have a handle on how to proceed.
We'll preserve energy when redirecting bass (AlexB's Sqrt(2) example), which is a little different than what we're doing now.
We'll then add an "avoid clipping" stage that attenuates all channels equally based on the maximum possible gain.
This raises a question: if you turn on room correction and just pick "+3 dB for the center speaker", is it confusing if all the other speakers turn down in volume instead of boosting the center? This would be the safe way to do it, but I think it makes calibration complicated.
We could also make the sliders only allow negative gains, but this makes it a pain to boost a speaker.
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From my reading, it sounds like it assumes either your amp or subwoofer will make the LFE 10 dB louder.
I have not done my reading yet, but since in this case MC is the preamp/processor I think it should perform the +10 dB trick unless the used decoder filter has already done it.
So that means when mixing from 2.0 to 2.1 with bass redirection, the LFE should be attenuated by 10 dB so it matches the levels you would get when playing a movie.
I suppose we could make additional options for "Move / Copy bass to subwoofer (LFE) at -0 dB".
Could the option be: "Adjust a separate LFE source +10 dB when it is mixed with redirected bass (needed to conform Dolby specs)"? Naturally you would also then need to prevent overflow.
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I still think that it would be good to have a way to adjust the balance of the redirected bass content and the original 0.1 LFE content freely. Following strictly the standard (or standards?) doesn't necessarily produce a well-balanced, enjoyable result.
Here's my old "submixer" graph:
(http://i238.photobucket.com/albums/ff132/alexb2k/MC/basssmall.png)
Here is an attempt to visualize its UI:
(http://i238.photobucket.com/albums/ff132/alexb2k/MC/SubwooferMixerUI.png)
However, perhaps the same could be achieved with a single LFE boost adjustment. You would still be able the adjust the master subwoofer level separately and thus fine-tune the balance:
(http://i238.photobucket.com/albums/ff132/alexb2k/MC/LFEboost.png)
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I still think that it would be good to have a way to adjust the balance of the redirected bass content and the original 0.1 LFE content freely.
You may be right. I think this whole area is complicated.
So when possible, it's good to make the program smart so the user doesn't need an audio engineering degree to get ideal sound.
Following strictly the standard (or standards?) doesn't necessarily produce a well-balanced, enjoyable result.
I am finding this to be true. The LFE level between popular television shows or different movies can be quite disparate.
I think putting the LFE gain on the OSD so I could adjust it from the couch with my remote would be nice.
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It looks like there has been some good progress on this over the weekend. I really don't know the answers to your earlier questions, Matt, but it sounds like to might have come up with someone else for the next build.
Bass management has caused problems for many companies. There is an old thread at AVSForum called LFE, subwoofers and interconnects explained (http://www.avsforum.com/avs-vb/showthread.php?t=748147) which pointed out the problems caused by receivers having to decode SACD, DVD-Audio, DVD, Blu-ray, and HD DVD and have consistent bass output levels.
Audioholics had an article called 0dBFS & Bass Management of DVD / Blu-ray Players (http://www.audioholics.com/education/audio-formats-technology/0dbfs-blu-ray). I just found it today, and it discusses the very issue of summing of channels and then shows how they tested the Oppo BDP-83SE. The measurement of the Oppo looks very similar to my Crossover +6dB chart above. My chart is showing two channels instead of one and I was using 24 dB/octave filters for both the High and Low Pass instead of 12/24 like Oppo uses.
(http://www.audioholics.com/education/audio-formats-technology/0dbfs-blu-ray/OppoBDP83SEFreq.jpg)
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It looks like there has been some good progress on this over the weekend. I really don't know the answers to your earlier questions, Matt, but it sounds like to might have come up with someone else for the next build.
A coming build will give you the flat curve you're looking for. We'll do an energy preserving accumulation of redirected bass.
We have to do a little internal plumbing first so that we know that the signal coming into Room Correcting is really 2.0 or 2.1 for music even though there are six or eight output channels.
There's still the issue of whether the LFE should be offset by 10 dB, but I think we'll start by making it possible to select a 0 dB or 10 dB shift since these are the only two "standards" I'm aware of.
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A coming build will give you the flat curve you're looking for.
I'm looking forward to seeing what a flat curve looks like. ;D
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We've got the updated algorithms mostly squared away. However, I have one remaining question:
Imagine playing 5.1 from a movie. There is redirected bass adding to the 0.1 from 5 channels.
To preserve energy, you would sum the five channels and multiply the sum by (1 / Sqrt(5)) or 0.45.
However, I'm not sure how this should combine with the existing 0.1. I can see three different arguments:
1) All six channels are the same -- sum the LFE and the 5 redirected bass values then multiply by (1 / Sqrt(6))
2) Combine the LFE and redirected bass in an energy preserving manner -- add the two values then multiply by (1 / Sqrt(2))
3) Simply add the LFE and the redirected bass together. No mixing done.
I think #3 is probably best or else the 0.1 channel would get attenuated as a result of redirected bass, which I don't think would be expected.
Any second opinions?
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I think receivers do number 3? They just take everything below crossover in all channels and send it to the sub. (after boosting the LFE 10 dB), so if you add bass to LFE from orther channels, it shoudl be attenuated 10 dB? BTW, afaik, music or movies shouldn't make a difference for a receiver, I think most receivers will boost the LFE-channel 10 db if receiving a 5.1 signal, so default should(?) to attenuate it by 10 db.
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I'd say #3.
A 0.1 source is a single source channel. One channel in => one channel out. No calculations for preserving its energy are needed.
The +10 extra gain is a separate issue. I've got the impression that typical HT receivers do it when Dolby* multichannel source with 0.1/LFE channel is decoded to PCM, but not when analog 5.1 or 7.1 signal is fed directly to the receiver's multi-channel inputs and digital signal processing is bypassed.
(* ...and perhaps also DTS, but I have not seen any documentation that would explain how LFE in DTS encoded signal should be handled.)
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I have been thinking about a situation in which low bass in the 5.0 audio content (i.e. not in the 0.1 channel) is mixed to come from a certain direction instead of mixing it more or less evenly to all five channels.
I created a sample in which only the front left channel of the five contains low bass (a combination of 30 Hz and 60 Hz sine waves). The audible volume level of this content should be preserved when it is redirected, but I don't think that will happen when the equation is (1 / Sqrt(5)).
I added also 0.1 aka LFE content to the sample (the same signal as above), but as already said, this part should be a no-brainer. One LFE in => one LFE out, except if the optional +10 dB gain is applied to this signal.
The 3 second sample in the attached zip package is in the 5.1 FLAC format.
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I have been thinking about a situation in which low bass in the 5.0 audio content (i.e. not in the 0.1 channel) is mixed to come from a certain direction instead of mixing it more or less evenly to all five channels.
I created a sample in which only the front left channel of the five contains low bass (a combination of 30 Hz and 60 Hz sine waves). The audible volume level of this content should be preserved when it is redirected, but I don't think that will happen when the equation is (1 / Sqrt(5)).
I added also 0.1 aka LFE content to the sample (the same signal as above), but as already said, this part should be a no-brainer. One LFE in => one LFE out, except if the optional +10 dB gain is applied to this signal.
The 3 second sample in the attached zip package is in the 5.1 FLAC format.
Thanks. What are you recommending?
A simple stereo addition example looks like this:
L = 1 volt = 1^2 units of power = 1 unit of power
R = 1 volt = 1^2 units of power = 1 unit of power
Simple addition: L + R = 2 volts = 2^2 units of power = 4 units of power
Power preserving addition: Sqrt(2) * (L + R) = 1.4 volts = 1.4^2 units of power = 2 units of power
Doesn't this mean, by definition, that if we attenuate to preserve power (not voltage) we would attenuate any single channel? Is there something we're missing?
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Alright, I think I have it. Let me know if this makes sense:
5.0 source, redirecting bass to the 0.1 channel (assume 0.1 channel is silent for now) should use this equation:
L^2 + R^2 + C^2 + SR^2 + SL^2 = LFE^2
Assuming values from 0 to 1.0, the maximum LFE is then Sqrt(5) or 2.24. This means to avoid clipping, all values must be multiplied by (1 / 2.24) or 0.45.
So, in AlexB's example of a single channel with full intensity sound:
L^2 + R^2 + C^2 + SR^2 + SL^2 = LFE^2
1.0^2 + 0^2 + 0^2 + 0^2 + 0^2 = LFE^2
1 = LFE^2
LFE = 1
Then, to adjust for attenuation:
LFE = 1 * 0.45 = 0.45.
The original left speaker also has a range of 0 to 0.45 after the attenuation. This means the movement of the bass from AlexB's example results in no difference in voltage or power.
Does this all check out? Basically we need to do squared summing with the root at the end.
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I think you are correct. Naturally also the original (now high passed) channels will get the same attenuation.
A practical test for e.g. L + R could be:
When the output is measured with an SPL meter in a listening room which method will produce the same reading from the single speaker as from the two original speakers? (Assuming all speakers are identical full range speakers and the room is an anechoic chamber.)
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The +10 dB LFE gain issue may still need to be considered. I found some indications that actually the subwoofer (or its poweramp) should be set +10 dB louder and when bass is redirected the final resulting mix should be corrected by -10 dB. The article that I referenced earlier made me believe that the Dolby decoder adds the +10 dB gain, but that may not be the case after all.
Here are a few more links that may be helpful:
http://www.ultimateavmag.com/howto/805bass/index.html
http://www.avsforum.com/avs-vb/showthread.php?t=748147
http://www.hometheatershack.com/forums/home-audio-subwoofers/1736-bass-effects-downmixing-dd-stereo-spectrum-labs-information.html
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I think you are correct. Naturally also the original (now high passed) channels will get the same attenuation.
A practical test for e.g. L + R could be:
When the output is measured with a dB meter in a listening room which method will produce the same reading from the single speaker as from the two original speakers? (Assuming all speakers are identical full range speakers and the room is an anechoic chamber.)
Assuming uncorrelated sound from the speakers (which is the most corret assumption i guess), it would be having 3dB less sound power in the two speakers. That is half the sound power. So the "correct" thin is to just add togheter all the sound power into the LFE-channel. However as mentioned, it needs to be normalized (all channels, full bandwidth), to make sure that the LFE-channel dont get clipped.
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The +10 dB LFE gain issue may still need to be considered. I found some indications that actually the subwoofer (or its poweramp) should be set +10 dB louder and when bass is redirected the final resulting mix should be corrected by -10 dB. The article that I referenced earlier made me believe that the Dolby decoder adds the +10 dB gain, but that may not be the case after all.
Here are a few more links that may be helpful:
http://www.ultimateavmag.com/howto/805bass/index.html
http://www.avsforum.com/avs-vb/showthread.php?t=748147
http://www.hometheatershack.com/forums/home-audio-subwoofers/1736-bass-effects-downmixing-dd-stereo-spectrum-labs-information.html
A properly set up receiver getting digital in should boost the channel 10 dB by itself. So the LFE-channel needs to be 10 dB lower. Correcting it in the sub is not a good solution imho.
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Actually, I meant just that the subwoofer may need to be set 10 dB louder than the other speakers when all signal processing is bypassed and only the last "volume knobs" are in the use. It doesn't really matter where these "knobs" are. They can be inside MC, in other software after MC, or in external hardware. The important thing is that the redirected bass content may need to be adjusted by -10dB before it reaches the final speaker adjustments and before it is mixed with the separate LFE channel. (edited this a bit)
For instance, according to what I have read lately, the signal that is used in the automatic speaker calibration systems in HT receivers (like in the YPAO system in my Yamaha receiver) is actually 10 dB lower in the LFE channel. This will automatically adjust the final subwoofer output to be 10 dB louder.
When we are feeding the analog outputs more of less directly to the poweramps we still need to calibrate the speakers somehow. It might be advantageous to preset the volume knob on the subwoofer to a quite loud position.
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Some Blu-rays have 7.1 audio, but most still have 5.1. I choose to upmix using JRSS all 5.1 to 7.1, but I still listen to 2.0 as 2.1. There may be some that don't choose to upmix and will listen in 7.1, 5.1, and 2.1. For these situations, you need to start your calculations with redirected bass from 7 channels and use the maximum headroom available. You will then need to further attenuate the mains/surrounds for 5.1/2.1 to match the levels from the 7 channel calculations. This is necessary in order for a calibrated system to be played back at the same levels regardless of the how many channels are being played.
In practical use, though, I think I would rather have each mode (7.1, 5.1, and 2.1) use the maximum headroom. Those of us using bass management probably aren't using a receiver. We don't have our volume control calibrated for reference levels (at least I don't) and we can just adjust the volume to our liking.
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Actually, I meant just that the subwoofer may need to be set 10 dB louder than the other speakers when all signal processing is bypassed and only the last "volume knobs" are in the use. It doesn't really matter where these "knobs" are. They can be inside MC, in other software after MC, or in external hardware. The important thing is that the redirected bass content may need to be adjusted by -10dB before it reaches the final speaker adjustments and before it is mixed with the separate LFE channel. (edited this a bit)
For instance, according to what I have read lately, the signal that is used in the automatic speaker calibration systems in HT receivers (like in the YPAO system in my Yamaha receiver) is actually 10 dB lower in the LFE channel. This will automatically adjust the final subwoofer output to be 10 dB louder.
When we are feeding the analog outputs more of less directly to the poweramps we still need to calibrate the speakers somehow. It might be advantageous to preset the volume knob on the subwoofer to a quite loud position.
The signal strength used during calibration has no direct bearing to the final output, the calibration playes a sound at a given level, and calibrates the sub so the level at the microphone is right. At least in principle. You can calibrate to excatly the same lever using a 70 dB signal as a 80 dB signal, as long as the software actually knows what level the signal is played at. That being said playing the LFE-channel lower in calibration makes no sense to me, why would you do that? The whole point of the LFE-channel is that is louder, the max SPL-level is louder, because it gets boosted 10 dB by the receiver. (higher max spl traded against higher noise floor). Besides background noise is much higher at lower freqs, frankly it seems like a stupid design. If you have the pwer, its always best to calibrate at a the highest volume pracitcal, and its not like adding or subtracting 10 dB needs to be done in the calibration itself.
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I think we are speaking about the same thing in different words.
Regarding the automatic adjustment systems, I meant the built-in hard coded test signals, not something that is user adjustable.
Here is a quote from my "AVS-forum" link:
Introduction
Initially, I'm only going to consider LFE in a system without bass management.
The LFE channel was originally found on multi-channel film recording formats, such as Dolby Stereo 70 mm Six Track. The purpose was to make more room for loud explosions, rumbles etc. The normal sound channels were calibrated so they could produce up to about 105dB SPL of output. Pretty loud, but not quite enough for the sort of low-frequency effects that you feel as much as hear.
So, to increase the impact, they added an extra channel, dedicated to low frequency effects, to be sent to a dedicated subwoofer. And this channel was calibrated to play 10dB louder than the rest - it could output 115dB SPL of energy. And 10dB difference is quite a lot - it means a signal over 3 times the amplitude.
Calibration
This 10dB boost is achieved by calibration in the monitoring amplifiers at the studio; a full-scale LFE signal on the tape is set up to play 10dB louder than a full-scale signal on any other channel. The cost is increased noise - the channel has been turned up, so the general hiss and noise on the magnetic recording will also be 10dB louder. But this is not too much of a problem, as all sounds above 150Hz are filtered out anyway on playback. The channel is only handling low frequencies.
To reproduce the effect heard in the studio, cinema playback systems are also calibrated the same way - a 10dB boost is added to the LFE channel amplifier.
Calibration tests generally hide this 10dB difference; a basic pink noise test will contain a signal that should play at 75dB SPL for each channel. For the normal channels, that means a signal 30dB below full scale (105dB-30dB = 75dB), but for the LFE channel it is a signal 40dB below full scale (115dB-40dB = 75dB): 1/3 of the signal amplitude. If those test signals are used to calibrate the amplifier so they all output at 75dB SPL, then everything is set up okay. The LFE channel now has 10dB of extra headroom above the other channels and can produce the full 115dB SPL when required.
Interconnects
One important point is that the 10dB boost to the LFE is performed at the final amplification stage. The potential extra volume of the LFE channel could not pass through interconnects, analogue recordings or digital recordings without going over the specified limits by a factor of 3. This, indeed, is the whole raison d'ętre of the LFE channel. It's a channel where by convention everything is recorded 10dB low to make room for louder bangs.
So at all points in a system - analogue interconnects, PCM signals, encoded Dolby Digital - the LFE channel is "10dB too low". A 75dB SPL signal on the LFE channel will be 3 times, or 10dB, smaller than a 75dB SPL signal on the other channels. It will only sound correct when fed to an amplifier that amplifies LFE by 10dB more than it amplifies the other channels.
This 10dB difference is dictated by Dolby and DTS, and the international standard ITU-R BR.1384-1:
Quote:The LFE channel is recorded with a level offset of –10 dB. This offset is compensated for in the reproduction system, where the LFE loudspeaker has an acoustic output (within its low frequency passband) of +10 dB with respect to the other channels.
In the home
The LFE channel progressed from early analogue magnetic formats, to the digital channel in cinematic versions of Dolby Digital, and in turn to the home versions of Dolby Digital, DTS and MPEG Multichannel found on DVDs.
The rules remain the same: the encoded LFE signal needs a 10dB boost. In receivers with integrated decoders, this is handled automatically in the internals, and the user is generally unaware of it. It is only when we try to connect amplifiers to external decoders that things can go wrong. Many receivers do not offer the necessary boost for LFE coming in from an external decoder.
LFE level adjustment
Some receivers with integrated decoders do offer an adjustment for the LFE level - an option to NOT boost the LFE by 10dB. This is needed for some early music mixes using DTS; the studio mixers, being unaccustomed to 5.1 production, hadn't calibrated their LFE channel 10dB high. Thus the recorded LFE signal on those discs is at equal level with the other channels, and doesn't require boosting.
The unboosted LFE option on the receiver is often called "LFE -10dB", with the standard boosted option being "LFE 0dB". Alternatively the options might be called "LFE 0dB" and "LFE +10dB". Whatever, the higher option is correct for films and almost all DD and DTS multichannel music. Since those early errors, both Dolby and DTS have specifed that LFE in music releases should be mixed for a 10dB LFE boost to match film soundtracks. However, SACD still differs... more on this below.
The article continues by explaining bass redirection. It is a good read.
http://www.avsforum.com/avs-vb/showthread.php?t=748147
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Hey, Alex, I already linked to that thread in post #22. :P
The thread does have good information and points out why receivers boost the LFE by 10dB before combining with the redirected bass. A Guide to Bass Management (http://forum.blu-ray.com/subwoofers/95817-guide-bass-management.html) over at the Blu-ray forum also has some good information and charts on how receivers handle bass management. All charts show the LFE channel being boosted by 10dB before being combined with redirected bass.
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Hey, Alex, I already linked to that thread in post #22. :P
Oops...
... All charts show the LFE channel being boosted by 10dB before being combined with redirected bass.
Correct. (...or the redirected bass is attenuated by -10 dB and the combined signal is boosted - the result is the same.)
When MC handles all processing the following options are necessary:
LFE redirected
1. +10 0 default
2. 0 0 the content is mixed without taking the +10 dB boost into account
3. 0 -10 the boost happens after MC in some other component
4. -10 -10 the content is mixed without taking the +10 dB boost into account & the boost happens after MC in some other component
The #3 and #4 can be achieved by reducing the speaker volume by 10 dB so maybe they are not needed separately. In any case, if the +10 dB correction is applied outside MC in the audio HW then the clipping prevention adjustment can be less excessive.
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We've added this to a coming build:
NEW: Added DSP Studio > Output Format subwoofer option "Calibrated to +10 dB (Dolby standard)". (used by JRSS and also Bass Management)
If you're redirecting bass, I would recommend making sure something (amp or subwoofer itself) is boosting by 10 dB so you can enable this option. The extra headroom means we have to turn other channels down a lot less for clip protection.
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The +10 dB correction cannot take place outside MC for the LFE track if it is already combined with the redirected bass. The only way this could happen is if you were routing the redirected bass to a different output than the LFE channel. Once it is combined, there is now way to only adjust the LFE track. Movie theaters and the studio can do it because only the LFE track is going to the subwoofer.
I'm curious how downloaded 5.1 music, like from iTrax, is mixed. I know SACD did things different, but I don't think any one listens to an SACD on a computer. We may need option 1 & 2 or 3 & 4, but I don't think we need all four options. If DVD-Audio, DVD, and Blu-ray all attenuate the LFE track and downloaded music does, too, then we don't even need an option.
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We've added this to a coming build:
NEW: Added DSP Studio > Output Format subwoofer option "Calibrated to +10 dB (Dolby standard)". (used by JRSS and also Bass Management)
If you're redirecting bass, I would recommend making sure something (amp or subwoofer itself) is boosting by 10 dB so you can enable this option. The extra headroom means we have to turn other channels down a lot less for clip protection.
So it can handle the #1 and #3. Correct?
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I think the #2 and #4 are essential because:
From: http://www.hometheaterhifi.com/technical-articles/163-the-misunderstood-01-lfe-channel-in-51-digital-surround-sound.html
DTS's LFE channel in consumer applications, unlike its cinema counterpart, is discrete but still has a few 'special' considerations. In the early days of mixing 5.1 for DTS CDs, the studios were not being calibrated correctly, with the end result of the LFE channel being set too low. When the material is played back on a correctly calibrated system, the LFE channel is way too high. THX was forced to introduce DTS Music and DTS Movie playback modes which distinguish between a correct LFE setting and a -10 dB setting to compensate for material assembled under mis-calibrated circumstances (though not all THX processors offer this convenience).
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I think we are speaking about the same thing in different words.
Regarding the automatic adjustment systems, I meant the built-in hard coded test signals, not something that is user adjustable.
Yeah, but it is adjustable to the manufacturer, test tones should be a given level, (or a few levels), and they should be as loud as possible, having a low level on bass test tones is not wise.
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I think #2 and #4 are essential because:
From: http://www.hometheaterhifi.com/technical-articles/163-the-misunderstood-01-lfe-channel-in-51-digital-surround-sound.html
Hmm, let me think, i miscalibrated 5.1 track would have the same level on the LFE track as the rest, MC adds the extra data to the LFE track from other channels, so the relative bass level is ok betwenn normal channels and LFE, however, the LFE is still 10 db to high. The difference should maybe be wheter 10 db is subtractet BEFORE or AFTER the channels are added in to LFE. (before being the default with well behaved source material)
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I think #2 and #4 are essential because:
I just e-mailed AIX Records/iTrax to see if #2 & #4 are necessary for their 5.1 music mixes.
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The +10 dB correction cannot take place outside MC for the LFE track if it is already combined with the redirected bass. The only way this could happen is if you were routing the redirected bass to a different output than the LFE channel. Once it is combined, there is now way to only adjust the LFE track. Movie theaters and the studio can do it because only the LFE track is going to the subwoofer.
I'm curious how downloaded 5.1 music, like from iTrax, is mixed. I know SACD did things different, but I don't think any one listens to an SACD on a computer. We may need option 1 & 2 or 3 & 4, but I don't think we need all four options. If DVD-Audio, DVD, and Blu-ray all attenuate the LFE track and downloaded music does, too, then we don't even need an option.
Thats no problem, as long as channel bass is attenuated 10 dB before its added to the LFE track.
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So it can handle the #1 and #3. Correct?
The option attenuates redirected bass by 10 dB, which I think is #3 (which is functionally identical to #1). The option is enabled by default. With the option off, you get #2 and #4 (we don't know if that native LFE line is assuming a +10 dB gain).
From what I can gather, FFDShow and AC3Filter (two popular AC3 / DTS decoders) both assume there's an external +10 dB gain on the 0.1 they output. It may turn out to be best for us to just recommend (force) calibration that matches this assumption, so that we're not trying to accommodate too many permutations.
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BTW, subtracting 10 dB BEFORE the normal channels are added to LFE is important, if not relative bass levels will be wrong between the existing LFE--track and the new data added from the normal channels.
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I just received a response back from Dr. Mark Waldrup, the president of AIX Records (http://www.aixrecords.com/). He said their LFE track is attenuated by 10dB during the mix.
By the way, iTrax/AIX Records is to high resolution music as MC is to audio players.
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I just received a response back from Dr. Mark Waldrup, the president of AIX Records. He said their LFE track is attenuated by 10dB during the mix.
Thanks.
Are there _any_ mainstream sources that have an LFE track that aren't assuming a +10 dB boost by the amp / subwoofer?
It won't give you those "flat curves" mojave, but I think maybe we could just drop the option and always attenuate redirected bass by 10 dB? (the curves would be offset by 10 dB)
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Thanks.
Are there _any_ mainstream sources that have an LFE track that aren't assuming a +10 dB boost by the amp / subwoofer?
Not as far as i know, there are some examples of SACD and the like not doing it correctly, but I think as good as everything assumes a 10db boost. I think its wise to just assume 10 dB boost, and eventually create a DSP plugin that lets you control volume on all channels independantly, for people with special needs.
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and eventually create a DSP plugin that lets you control volume on all channels independantly, for people with special needs.
Just a little note to say the 'Room Correction' DSP already allows per-channel volume adjustments.
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Are there _any_ mainstream sources that have an LFE track that aren't assuming a +10 dB boost by the amp / subwoofer?
To be correct, I think most mainstream sources are assuming bass management will be used by the consumer in a DVD/Blu-ray player that is using analog outputs, receiver, or preamp/processor and that the LFE track will be boosted +10 dB.
Please read Alex's link from post 11. It is excellent for understanding the LFE channel. Here is the link again:
Feature Article - "The Misunderstood 0.1 LFE Channel in 5.1 Digital Surround Sound" (http://www.hometheaterhifi.com/volume_7_2/feature-article-misunderstood-lfe-channel-april-2000.html)
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Just a little note to say the 'Room Correction' DSP already allows per-channel volume adjustments.
Indeed it does, then I say, ditch all the other unusual settings, with the possible exception of not attenuating bass data from channels 10 dB before adding them to LFE, and if people have very special needs, they can use the per-channel volume adjustments. That should take care of most problems(?)
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Are there _any_ mainstream sources that have an LFE track that aren't assuming a +10 dB boost by the amp / subwoofer?
The quote I posted earlier mentioned the DTS Music CDs. They were never mainstream, but there are lots of them in use. I have a few.
You can still buy DTS Music CDs, but I don't know how the LFE channel is calibrated in the available discs:
http://www.5point1.com/Products-5_1_DTS_Music_Disc_2.html
some other links:
http://en.wikipedia.org/wiki/5.1_Music_Disc
http://www.dts.com/Pro_Audio_Products/Products_by_Media/DTS_Music_Disc_CD/Overview.aspx
http://www.avrev.com/equipment-reviews/music+disc+reviews/music+disc+reviews/dts+5.1+cd.html
I'd suggest to preserve the option so that the calibration problem can be fixed if needed.
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I'd suggest to preserve the option so that the calibration problem can be fixed if needed.
I think you had it right that there are really two questions:
[ x ] Something outside of Media Center will make the subwoofer channel 10 dB louder
[ x ] The data already in the LFE channel expects something to make it 10 dB louder
We added an option "Calibrated to +10 dB (Dolby standard)" under Subwoofer in DSP Studio > Output Format for the first question, but I think you're talking about the second question.
Any suggestions as to option names if we do want it to be optional?
Subwoofer
[ x ] Calibrated to +10 dB (Dolby standard)
[ x ] LFE soundtrack expects +10 dB
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I think you had it right that there are really two questions:
[ x ] Something outside of Media Center will make the subwoofer channel 10 dB louder
[ x ] The data already in the LFE channel expects something to make it 10 dB louder
We added an option "Calibrated to +10 dB (Dolby standard)" under Subwoofer in DSP Studio > Output Format for the first question, but I think you're talking about the second question.
Any suggestions as to option names if we do want it to be optional?
Subwoofer
[ x ] Calibrated to +10 dB (Dolby standard)
[ x ] LFE soundtrack expects +10 dB
I dont really see the difference the way you have worded it, but I might just misunderstand you. The important question is this, is the existing LFE in the material 10 db lower than the rest of the channels? This decides if MC should mix in added content to LFE-channel 10db lower or not. The handling outside of MC is easily fixed by just dropping the voulme on the subwoofer channel, the important is that LFE-content and "the rest of the channels"-content mixed into LFE-channel are at the right level relative to each other.
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The handling outside of MC is easily fixed by just dropping the voulme on the subwoofer channel
For some users, this is true.
But for others, there's no such thing as "handling outside". I only have a power amp.
So to play 5.1 material mastered different ways, the program needs to know how the material was mastered and how the stereo is configured so that it can deliver consistent results.
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For some users, this is true.
But for others, there's no such thing as "handling outside". I only have a power amp.
So to play 5.1 material mastered different ways, the program needs to know how the material was mastered and how the stereo is configured so that it can deliver consistent results.
Looks like there will be three options to Calibrate the LFE:
1) MC
2) Receiver (which Matt does not have)
3) Vol Control on the Sub itself (which Matt is going to make!)
I'm looking forward to hearing these changes (as I thought the bass overemphasised in the 5.1 mix) once my HT is back together. As usual I plan to calibrate my 5.0 speakers (using a Sound Pressure Meter and test tones) via the Receiver and then bring the Subs levels up to match using the Volume Control on the Sub itself.
Thanks
Nathan
PS - I'm also hoping that these changes don't make MC's output levels different with other HTPC apps, eg TMT which I use for my blu-ray playback, or other sources such as the STB as then there becomes nother layer to calibrate to bring the entire HT into balance!
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The questions:
[ x ] Something outside of Media Center will make the subwoofer channel 10 dB louder
[ x ] The data already in the LFE channel doesn't expect something to make it 10 dB louder
(in other words: the poor guys in the studio didn't have their subwoofer correctly set to +10 dB and the recording sounds good only if also the playback system doesn't boost LFE)
The options:
Subwoofer
[ x ] Already calibrated to follow the Dolby/DTS +10 dB LFE standard (the internal LFE gain will be disabled)
LFE input
[ x ] LFE soundtrack doesn't expect +10 dB (use only if the recording does not follow the standard)
EDIT: changed Dolby/DTS to +10 dB LFE
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I agree with Elvis133's entire post.
Here is what I think you might be getting at, but I'm probably misunderstanding, too:
1. You are mixing the redirected bass at -10 dB to match the LFE. This is then being sent out and the subwoofer is set at a higher level to match the rest of the channels. This allows more headroom for the other channels.
2. You are mixing the redirected bass at -10 dB to match the LFE channel. This is then increased by 10 dB (or other channels attenuated) so that all channels including are output at the same level. This is how all receivers do it.
I'm not sure why you are referencing the Dolby standard because it isn't just Dolby that does this. Whether a Blu-ray outputs in Dolby TrueHD, DTS-HD, or PCM, the LFE track has still been attenuated by 10dB in the studio. This has become the standard for all multi-channel mixing except for the rare case of some DTS audio and some SACD's. Probably these are even mixed to this standard now.
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The issue isn't about how the subwoofer is calibrated, it is whether or not the LFE track has been attenuated in the studio by 10 dB. The two options should be:
LFE & Redirected Bass Mixing
1. LFE track is -10dB per normal mixing standards (DVD, DVD-Audio, Blu-ray)
2. LFE track is the same as other channels (some DTS Audio and SACD)
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The issue isn't about how the subwoofer is calibrated, it is whether or not the LFE track has been attenuated in the studio by 10 dB. The two options should be:
LFE & Redirected Bass Mixing
1. LFE track is -10dB per normal mixing standards (DVD, DVD-Audio, Blu-ray)
2. LFE track is the same as other channels (some DTS Audio and SACD)
The two options can be set by the latter tickbox:
LFE input
[ x ] LFE soundtrack doesn't expect +10 dB (use only if the recording does not follow the standard)
The subwoofer calibration is a different issue. Apparently Matt would like to provide an easy setting for it. The first tickbox would do the trick:
Subwoofer
[ x ] Already calibrated to follow the +10 dB LFE standard (the internal LFE gain will be disabled)
These two tick boxes will actually provide the funtionality of my # 1, 2, 3 and 4.
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I posted the above so it could be read while typing some more.
There are two ways #1 could be handled which was what I posted earlier:
1. You mix the redirected bass at -10 dB to match the LFE. This is then being sent out and the subwoofer is set at a higher level to match the rest of the channels. This allows more headroom for the other channels.
2. You are mixing the redirected bass at -10 dB to match the LFE channel. This is then increased by 10 dB (or other channels attenuated) so that all channels including are output at the same level.
If you have a subwoofer with a pro amp or no volume control then #2 might be necessary. If you have volume control in your amp then #1 is preferred. This brings us to the following options:
LFE & Redirected Bass Mixing
1. LFE track is -10dB per normal mixing standards (DVD, DVD-Audio, Blu-ray)
a. Summed bass is -10 dB for maximum channel headroom
b. Summed bass is the same level as other channels
2. LFE track is the same level as other channels (some DTS Audio and SACD)
However, 1b can be accomplished by just using the subwoofer gain control in Room Correction. You would never just output directly to an amplifier without some volume compensation because of varying driver efficiencies. I don't think the equal levels is necessary and 1a would be fine as the option. Now we are back to:
LFE & Redirected Bass Mixing
1. LFE track is -10dB per normal mixing standards (DVD, DVD-Audio, Blu-ray)
2. LFE track is the same level as other channels (some DTS Audio and SACD)
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For some users, this is true.
But for others, there's no such thing as "handling outside". I only have a power amp.
So to play 5.1 material mastered different ways, the program needs to know how the material was mastered and how the stereo is configured so that it can deliver consistent results.
Then you can just adjust the LFE-level in the room correction DSP as long as the relative levels of material added to LFE, and the original LFE-channel is the same.
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However, 1b can be accomplished by just using the subwoofer gain control in Room Correction. You would never just output directly to an amplifier without some volume compensation because of varying driver efficiencies. I don't think the equal levels is necessary and 1a would be fine as the option. Now we are back to:
LFE & Redirected Bass Mixing
1. LFE track is -10dB per normal mixing standards (DVD, DVD-Audio, Blu-ray)
2. LFE track is the same level as other channels (some DTS Audio and SACD)
I agree 100%, those are the only two options we need, the LFE-bass material is either 10 db lower or the same as the bass-material in the 5 other channels, as long as this material is added togheter correctly, the absolute level of the LFE-channel is much easier to fix/modify later. If the bass is not mixed together properly, it cannot be fixed by a simple gain adjustment in MC, receiver, sub or other part of the chain.
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Alex, I guess I am breaking down your LFE option into two options instead of an on/off. This is just to help make it more clear to the user.
I see how the Subwoofer option might be necessary, but anyone that is using other sources and is calibrated to those will be astute enough to make sure the sub is calibrated in MC, too, and just use the levels control in Room Correction.
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Here is how the four options can be set:
1.
LFE redirected
+10 0 could be the default. Useful when you have a calibrated HT receiver, but the digital calibration is bypassed when the analog 5.1 inputs are used.
Subwoofer
[ ] Already calibrated to follow the +10 dB LFE standard (the internal LFE gain will be disabled)
LFE input
[ ] LFE soundtrack doesn't expect +10 dB (use only if the recording does not follow the standard)
2.
LFE redirected
0 0 the content is mixed without taking the +10 dB boost into account
Subwoofer
[ ] Already calibrated to follow the +10 dB LFE standard (the internal LFE gain will be disabled)
LFE input
[ x ] LFE soundtrack doesn't expect +10 dB (use only if the recording does not follow the standard)
3.
LFE redirected
0 -10 the boost happens after MC in some other component. Provides louder output and more dynamic range (to all channels) if you can set the subwoofer permanently to +10 dB outside MC
Subwoofer
[ x ] Already calibrated to follow the +10 dB LFE standard (the internal LFE gain will be disabled)
LFE input
[ ] LFE soundtrack doesn't expect +10 dB (use only if the recording does not follow the standard)
4.
LFE redirected
-10 -10 the content is mixed without taking the +10 dB boost into account & the boost happens after MC in some other component
Subwoofer
[ x ] Already calibrated to follow the +10 dB LFE standard (the internal LFE gain will be disabled)
LFE input
[ x ] LFE soundtrack doesn't expect +10 dB (use only if the recording does not follow the standard)
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It sounds to me like everyone is in agreement, and only haggling over naming. That's good.
I was going to ask if anyone would ever not check the option "Something outside of Media Center will make the subwoofer +10 dB". Then AlexB recommended not checking it by default and gave a good reason -- analog paths may bypass the boost of a calibrated receiver.
Thanks everyone.
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I have a proposal for how to do the user interface. In DSP Studio > Output Format, add a root level section 'Bass Management'. Remove 'Subwoofer' settings from 'Channels'.
The new section would look like this:
=== Bass management ===
[ x ] Something outside Media Center will make the subwoofer +10 dB (standard calibration)
* When playing sources with no LFE (CD audio, etc.)
[ Crossover combobox (Silence, No crossover, 20 Hz, 25 Hz, etc.) ]
[ x ] Subclarity (tm) for cleaner, tighter subwoofer output
* When playing sources with an LFE (DVD, HDTV, etc.)
[ x ] LFE is correctly mixed to +10 dB (only uncheck if the recording does not follow the standard)
Bass management is used to control how the subwoofer is handled. The settings only apply when the output format has a subwoofer channel.
You can redirect bass to the subwoofer using the 'Room Correction' DSP.
By default, all checkboxes would be checked.
This way, by default, no audio changes would be needed when playing a 5.1 or 7.1 source. Both the system and the LFE would be at the same level.
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One final question: how technical should we be with regards to the terminology "subwoofer" vs "LFE"?
My inclination is to just always say "subwoofer" because I think it's easy to understand. My wife does not know what an LFE is, but she knows that a subwoofer is that annoying thing behind the couch.
But I'm afraid this less-than-technically-pure usage may offend some people. Advice?
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Well, I finally understand why the electronic measurement should not be flat because of preserving energy in the satellite channels.
I really like the idea of having a bass management DSP. However, I still don't like the idea of the crossover combo box (silent, no crossover, 20 Hz, etc.). It only confuses things with the crossover settings in Room Correction. You can still do all this with the settings in Room Correction without someone creating double crossovers and filtering out part of their music. You (Matt) mentioned before that you like a different setting for music.You still need to turn off the crossovers in room correction before enabling them in the bass management DSP (or Output Format). It really isn't saving anything. You could have two zones setup: one for music and the other for movies. You could even make an option in the Video settings to use the Video Zone.
I think the Bass Management DSP should have some default settings that are used by room correction (and it should be enabled whenever Room Correction is enabled). These settings would be the crossover frequency for each speaker (same for all speakers); whether the redirected bass is moved, copied, or removed; and the High and Low Pass Crossover slopes. It gets very tedious to change things all the time in Room Correction for each channel - at least when trying different stuff and calibrating the system.
In Room Correction you just add "Default" as an option to each of the above mentioned items. This allows for global changes in Bass DSP and variations in Room Correction.
Regarding the use of LFE and Subwoofer, I think you are using them correctly in the above bass management rough draft. I would actually add the word "channel" after LFE and call it the LFE channel. I thnk it is best to be technical. It makes it better in the long run when consistent terminology is used among varous manufacturers, magaines, software, etc.
Good Night!
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However, I still don't like the idea of the crossover combo box (silent, no crossover, 20 Hz, etc.). It only confuses things with the crossover settings in Room Correction.
A 5.1 source has LFE and redirected bass, so why wouldn't 2.0 upmixed to 5.1 (or 2.1) have an LFE and redirected bass? It seems like consistently structured audio coming out of 'Output Format' (always the first DSP) is desirable and makes getting consistent results from Room Correction easier. The only counter-argument I can think is that the amount of redirected bass might be less for a 5.1 source depending on how they filtered it during mastering.
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A 5.1 or 7.1 source does not have redirected bass. It only has the LFE channel and you are handling that in Room Correction and the Bass Management DSP. Dolby Pro Logic II or IIx only creates a 5.0 or 7.0 mix from a 2.0 source (and DTS:NEO 6 creates 5.0 or 6.0). It does not make an LFE channel. The LFE channel is created from redirected bass by the bass management used by the receiver. You used to not have bass management so JRSS had to create the .1 channel. JRSS should now do the same as Dolby Pro Logic to prevent problems with multiple crossovers. Have JRSS create 4.0, 5.0, or 7.0 mixes from 2.0 content. For 5.1 content, have it just create the surrounds and leave the LFE alone (like it does now). Then let Room Correction/Bass Management determine the crossover, etc.
Like I said before, you can't use both settings at the same time and Bass Management accomplishes the goal in all circumstances.
Dolby Surround Pro Logic II Decoder Principles of Operation (http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/209_Dolby_Surround_Pro_Logic_II_Decoder_Principles_of_Operation.pdf)
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A 5.1 or 7.1 source does not have redirected bass. It only has the LFE channel and you are handling that in Room Correction and the Bass Management DSP. Dolby Pro Logic II or IIx only creates a 5.0 or 7.0 mix from a 2.0 source (and DTS:NEO 6 creates 5.0 or 6.0). It does not make an LFE channel. The LFE channel is created from redirected bass by the bass management used by the receiver.
AFAIK that is not entirely precise, the receiver takes all material below crossover and outputs it to the sub, no matter if its in the LFE-channel or any other channels, I don't think it creates a LFE-track per se if it gets 5.0 input.
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I'm looking forward to testing all this on the WE, I plan to:
1) Recalibrate the 5.1 speakers from the reciever's test tones using a SPL meter
2) Check calibration output from MC using it's test tones
Thanks
Nathan
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A 5.1 or 7.1 source does not have redirected bass.
Are you saying you want bass redirection (and speaker crossovers?) disabled when playing 5.1 or 7.1 content? I think this is the first I've heard of this, but maybe it could be an option.
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I think his point is that you should first create the full range channels from the source signal and only after you have the finished new audio signal ready apply the bass management.
If a separate LFE channel exist in the original unmodifed source it should be kept out of all processing except the +10 dB adjustment that will be applied only to the LFE signal (if enabled).
Naturally in the case of the separate LFE source you must maintain the correct balance between the volume levels of the other channels and the LFE channel (in this equation the LFE channel is normally set to be +10 dB louder than it was in the original source data.)
So, for instance, when you create surround from a stereo source create only the five (or seven) full range channels and only after that let the bass management determine if the lowest frequencies should be redirected to the subwoofer. (Isn't this already how it works?)
Another example could be 5.1-> 7.1:
1. Process the five full range channels and create seven channels.
2. Manage bass, i.e. create the redirected bass mix and the seven high passed channels
3. Mix LFE (+10dB) and the redirected bass mix (0 dB) and create the subwoofer channel
Which is exactly the same as this:
(http://i238.photobucket.com/albums/ff132/alexb2k/MC/basssmall.png)
- except that the subwoofer mixer is not implemented in the UI as a mixer with adjustable sliders. The only available adjustment is the on/off switch for the +10 dB gain so that incorrect recordings can be fixed.
Here's one more step (not pictured):
4. A setting that attenuates the final subwoofer channel by -10 db if:
"Something outside Media Center will make the subwoofer +10 dB"
This is the logically correct order of the steps. Naturally you must decide how the processing is actually done under the hood. You may need to change the order, combine some steps and adjust the gain in several places in order to maintain the correct balance and prevent the final output from clipping. As long as the end result is the same that doesn't matter.
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For clarity, I'm going to start calling the output that contains the LFE and redirected bass "Channel 4." This is the actual channel number that is being used. On Channel 4, you might have an LFE track which is the Low Frequency Effects (LFE) created in the studio for movies, you might have a bass track on multichannel audio, and it will be blank with a stereo source. Channel 4 is also used for the redirected bass from other channels.
Are you saying you want bass redirection (and speaker crossovers?) disabled when playing 5.1 or 7.1 content? I think this is the first I've heard of this, but maybe it could be an option.
No, I don't want that at all. I was responding to your statement that "A 5.1 source has LFE and redirected bass." I was trying to say that a 5.1 source has all discrete channels and the redirected bass is the function of bass management. I was also pointing out that Dolby and DTS matrixing leave Channel 4 blank when matrixing. It is the bass management that either copies or moves the redirected bass to Channel 4.
You aren't really "creating" anything on the channel 4 with JRSS 5.1 or 7.1. You are leaving the other channels full range, and copying the bass below the crossover to channel 4. This is the same as using Bass Management and setting the routing in Room Correction to "Copy bass to Subwoofer (LFE)." By having it done in JRSS, you are removing the ability to set High and Low Pass Filter slopes. You are also causing confusion because some will set the crossover both in JRSS and in Bass Management. This will result in problems with the frequency response and phase.
Try this: Listen to a stereo audio track with JRSS on with 5.1 output and Subwoofer to 100 Hz and disable the bass routing for the L & R channels in Room Correction.
Now set the JRSS's Subwoofer setting to silent and in Room Correction set the routing for the L & R channel to a crossover of 100 Hz with "Copy bass to Subwoofer" selected. Set the slopes to match whatever you use when the crossover is active in JRSS. Now listen again. These should both sound exactly the same. Below are what they look like.
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The High and Low Pass Crossovers in Room Correction need to be labeled. I know what they are, but most people wouldn't be able to tell them apart. They also shouldn't be selectable in Room Correction: Subwoofer.
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To continue my previous reply:
A different and very common case would be
5.1 -> 2.0
According to the links in this thread most HW players simply drop the LFE channel from the mix when the stereo analog outputs are used. This may be good if the speakers are not up to the task of reproducing the low frequency effects.
However, the user may have an standard analog stereo system with a good subwoofer (or perhaps just a pair of huge full range speakers). It might be good to provide an option to include LFE in the mix. (The channel mixers in the AC3filter and FFDSHow decoders have this option, but not all 5.1 formats are decoded through DirectShow.)
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I think it's correct to allow the option, like we do now, to fill the subwoofer on a 2.0 to 7.1 upmix using JRSS or to leave it silent so a user can use only redirected bass.
A native 7.1 mix would have information in the LFE channel, so it seems reasonable for JRSS to do the same when simulating 7.1.
This option also makes it straight-forward for a user to select 7.1 output and get something good on all the channels. They could later explore the more complicated Room Correction with bass redirection.
I also believe that bass management, using redirected bass, is not going to be able to build as good of a subwoofer output from a 2.0 to 7.0 JRSS mix as JRSS itself could have built. If bass redirecting did not redirect simulated channels (so 'Move bass' would become 'Remove bass' for simulated channels) it might be better. But then it would be acting differently than configured, which could be a problem.
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I think it's correct to allow the option, like we do now, to fill the subwoofer on a 2.0 to 7.1 upmix using JRSS or to leave it silent so a user can use only redirected bass.
If you are going to allow JRSS to create a bass channel when upmixing, then you should, at the very least, have this crossover disabled when someone uses bass management.
A native 7.1 mix would have information in the LFE channel, so it seems reasonable for JRSS to do the same when simulating 7.1.
No other matrixing method makes a .1 channel because of the problems it causes when Bass Management is used.
I also believe that bass management, using redirected bass, is not going to be able to build as good of a subwoofer output from a 2.0 to 7.0 JRSS mix as JRSS itself could have built.
I disagree for several reasons. First, JRSS assumes that your speakers are full range so it doesn't move bass, but just copies it. Now one has not just two, but 5 or 7 channels of full range audio. 30%- 40% of the power of the amplifier is used for the bass frequencies. This is putting an extra load on the amplifier and it will clip a lot sooner. Also, most receiver's maximum output is rated for two channels. When the surround channels are in use, the available watts can drop quite a bit. Having all the frequencies playing will cause the drivers in the speaker to distort at lower volumes.
Second, JRSS only uses a 12 dB/octave slope on the low pass filter (at least that is what I measure). The standard for all receivers is a 24 dB/octave slope. Have such a shallow slope means that you will be playing out of the linear range of the of subwoofer reducing sound quality. It also makes the subwoofer more directional. In other words, it makes it easier to hear where the subwoofer is located.
Third, having the bass in 6-8 sources makes it very difficult to calibrate and eliminate room modes and phasing problems. This is why the default for a receiver's room calibration software such as Audyssey or YPAO is to redirect all bass to the subwoofer using Bass Management. It is much easier to EQ one source that multiple sources.
Fourth, have the bass in 6-8 sources that might vary in quality will mean that your bass is only as good as the worst speaker. For example, someone might have nice mains, center, and even subwoofer, but have crappy surrounds. The muddy bass coming out of the surrounds will mask the other bass and reduce its quality.
There are good reasons one might want full range bass in all speakers, but for JRSS to do this and be considered the beginner's option (with Bass Management being more advanced) is a little backward. Routing all bass to Channel 4 is the beginner's option and will produce the best quality in the most systems. Choosing full range audio in all channels or a 12/db slope on the low pass filter are more advanced settings/options that require better amplification, better drivers, and a better understanding of calibration in order for it to work well in someone's room. 8)
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I disagree for several reasons.
You've made some very good arguments. Thank you.
One little note is that JRSS is using a 48 dB/octave slope for the subwoofer since build 15.0.74. I just measured it and it seems to be working correctly.
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I also believe that bass management, using redirected bass, is not going to be able to build as good of a subwoofer output from a 2.0 to 7.0 JRSS mix as JRSS itself could have built.
I can understand this because the signal in the surround channels probably isn't in the original phase. If you combine bass after creating the surround channels the signals in the opposite phase will cancel each other out (more or less).
I suppose you could filter and combine the bass content according to the bass management settings from the stereo signal and only after that create the other channels and apply the high pass filter (if bass management is set to do so).
In any case, the fake surround mode is not important to me. I normally prefer to have the original channels.
Correctly implemented bass management is a lot more important thing to have. I have not even hooked up the analog 5.1 outputs yet (except some experiments years ago), but I will need the bass management features when I do it. I have a growing number of multi-channel media files that don't contain Dolby or DTS audio and thus cannot be played through SPDIF (without re-encoding to AC-3). My receiver can apply bass management only to digital signals.
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I did some real world testing tonight. I'll start by saying I'm only trying to achieve sound I like, not sound that is correct according to some text-book definition. I appreciate both neutral bass and exaggerated bass. Below are my personal opinions with regards to the results:
I was testing with stereo files played on a 5.1 system, using all the speakers.
First, I found the highest crossover points for my speakers at which I couldn't hear any audible effect from the crossover. This seems like the logical place to crossover the speaker since any energy below that frequency would be wasted (and possibly distort the other sound).
It worked out to:
L / R: 40 Hz / 24 dB/octave
C: 80 Hz / 24 dB/octave
SL / SR: 100 Hz / 24 dB/octave
Next, I experimented with how to get the best sounding subwoofer output.
I used the two main approaches we've been discussing:
1) JRSS (using 'Remove bass' in bass management)
I simply love the way this sounds. Media Center + JRSS + Room Correction delivers the best sound I've ever gotten from any stereo.
I'm able to dial in a very clean subwoofer output in JRSS. This is especially good with rock music, where it does a nice job of keeping clean bass drum hits on the subwoofer, while keeping drone from bass guitars off the subwoofer.
2) Redirected bass only (using 'Move bass' in bass management)
This approach is energy neutral, so there's a mathematical purity to it. But I just couldn't get a bass output that sounded as good as JRSS. Using the crossovers established above, it put too much high bass on the subwoofer. Instead of tight punches, it droned and made bass guitars slide around the sound stage.
Moving sound below the 100 Hz crossover from the SL and SR speakers puts frequencies I don't think sound good on the subwoofer because of directionality, inability to exaggerate bass without exaggerating unwanted drone, etc.
Allowing a final (optional) subwoofer crossover might solve some of my problems, although then it's not clear the advantage of using #2 instead of #1.
This is all relevant because we're trying to figure out what to make most prominent in the user interface and what to recommend. From my testing, the best sound was with letting JRSS fill the subwoofer and using bass management to remove unsupported frequencies (#1 above).
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I disagree for several reasons. First, JRSS assumes that your speakers are full range so it doesn't move bass, but just copies it. Now one has not just two, but 5 or 7 channels of full range audio. 30%- 40% of the power of the amplifier is used for the bass frequencies. This is putting an extra load on the amplifier and it will clip a lot sooner. Also, most receiver's maximum output is rated for two channels. When the surround channels are in use, the available watts can drop quite a bit. Having all the frequencies playing will cause the drivers in the speaker to distort at lower volumes.
Ok, here I am getting confused, JRSS copies bass instead of moving it? Why would it do that? That woud create a lot more energy at the frequencies copied, and make the mix much more bass-heavy (wich have been a problem with JRSS, could this be the explanation?)
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Ahhh what fun! I think most of us will find the "correct" settings will be a mix of personal preference with the equipment we have (and the effort needed to go to smooth out any issues with the speakers). Looking at the frequency curves of my speakers I see that may front speakers start to roll off at 80Hz, with my sub flat from 18-90hz. Looks like the THX recommended cross over of 80hz suits my setup well.
That said I personally find the following listening scenarios:
1) "Sweet Spot" - Occasionally (very occasionally) I will sit in the sweet spot to listen to 2.0ch audio and much prefer to sound stage of a "pure direct" 2.0 output. I find any of the DSP (either the receiver or JRSS muddies the sound stage when upmixing to 5.1 or even (though to a lesser extent) 2.1 with bass redirection).
2) Casual Listening - Much more common is just "cranking it up" and listening while moving around. I really like the 5.1 upmix for this scenario and think the JRSS provides a great option to do this to ONLY my 2.0 stuff unlike the receiver that requires you to manually change the listening mode. That said JRSS to my ears was bass heavy in it's operation.
While I like this option for 2.0 material, I'll never use it for original 5.1 encoded material - I'm just happy with the mix done by the studio and don't feel the need to play with it.
Thanks
Nathan
PS - Matt, I did not think you had a sub?
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Yeah, jmone, i have it more or less the same, i use 5.1 source as it is, and upmix 2.0, however I have also had problems with JRSS being to bass-heavy. The upmixer should keep the bass energy the same as in the source, do it correct, and le people play with an EQ or similar if they want a more bass-heavy mix than the source.
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While I like this option for 2.0 material, I'll never use it for original 5.1 encoded material - I'm just happy with the mix done by the studio and don't feel the need to play with it.
You may know this, but if your output is set to 5.1 and your source is 5.1 and you have JRSS on, it doesn't do anything to the original mix. However, if output is set to 7.1 and the source is 5.1, it will matrix the rear surrounds. The only time JRSS does anything to Channel 4 (LRE/Subwoofer) is when the source has no info on that channel.
I have 7.1 output, JRSS on, "For stereo sources, only mix to 2.1" checked, and the crossover set to silent. This setting doesn't touch 2.0 or 7.1, but it matrixes 5.1 to 7.1 by adding the rear surrounds. Using Bass Management, I listen to 2.1 for music (if it is a stereo source) and 7.1 for movies.
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I did some real world testing tonight.
Somehow I missed this post. I see how JRSS could be producing better bass because it is probably working with just the original channels to create the subwoofer output. I just did some listening myself using JRSS and "Remove Bass" in Room Correction. It is very nice. However, when I want to watch a movie I have to change all the setting for each channel back to "move bass" or create two different zones with different settings and then instruct the family on how to change zones, etc.
Is there a difference between have JRSS create the subwoofer output or having Bass Management create the subwoofer output when I am only outputting to 2.1? If they are different then you could have the bass management work differently when a 2.0 source is expanded to 2.1, 5.1 or 7.1 by JRSS. Otherwise, you would only need to make a change when JRSS is upmixing to 5.1 or 7.1.
Maybe this would work. If the user is going to use bass management for movies, they will want to always leave their settings the same. However, like you pointed out, it might be best to use JRSS in some instances. The following then needs to happen:
1. The only place to setup crossovers and bass management is in Room Correction.
2. When JRSS is enabled and there is no subwoofer channel, then it looks at the crossover setting for the L & R (it would be nice if these were linked so changing one would change the other) and uses this crossover setting to send bass to the subwoofer channel using the Low Pass filter slope selected by the user. Contrary to regular bass management, it ignors the other channels when mixing to 5.1 or 7.1.
3. If Routing in Room Correction is set to "Remove" or "Move" JRSS automatically uses "Remove" since the bass below the crossover is already being sent to the subwoofer. It uses the user selected High Pass slope. If it is set to "Copy" then JRSS still sends the fullrange signal to the speakers.
This integrates JRSS with the Bass Management and provides the best of both without any user intervention required for differences between movies and music. It allows users to use the high and low pass filters of their choice. It only requires the crossover be set in one place. It still follows the JRSS mixing method for going from 2.0 to 2.1, 5.1, and 7.1. It allows the user to set everything once and not have to make changes depending on the source.
The only thing it actually changes in JRSS is allowing the Low Pass filter slope to be varied rather than set at 48dB. 8)
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Matt's done a good job and I think I've got my Bass where it should be - and it could have been multiple devices applying DSP / config issue on my part (and/or JRSS overemphasis?) - http://yabb.jriver.com/interact/index.php?topic=58771.0 Who knows but it is sounding pretty good to my ears.
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This integrates JRSS with the Bass Management and provides the best of both without any user intervention required for differences between movies and music. It allows users to use the high and low pass filters of their choice. It only requires the crossover be set in one place. It still follows the JRSS mixing method for going from 2.0 to 2.1, 5.1, and 7.1. It allows the user to set everything once and not have to make changes depending on the source.
How about a checkbox in Output Format > Subwoofer > When source has no subwoofer > - Disable redirected bass from Room Correction
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How about a checkbox in Output Format > Subwoofer > When source has no subwoofer > - Disable redirected bass from Room Correction
I would have thought that "Room Correction" would be the only place you needed to config all the Speaker Settings including for the Subwoofer for your particular environment. Output format would then just determin what channel mixing, sample rates and bitdepth you would use etc.
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How about a checkbox in Output Format > Subwoofer > When source has no subwoofer > - Disable redirected bass from Room Correction
That still doesn't address the issue of the user needing to change from "remove bass" to "move bass" in Room Correction depending on whether or not the "source has no subwoofer."
You could do this behind the scenes when the source has no subwoofer. I don't even see the need for a checkbox.:
1. Disable redirected bass from room correction.
2. Use the crossover setting from front left speaker in Room Correction for JRSS.
3. Get rid of the "remove bass" option. It is only needed in this scenario with JRSS. When "move bass" is selected and JRSS is creating the subwoofer output, then silently "remove bass." Otherwise, copy bass if the user has selected "copy bass."
4. This might be more difficult to implement, but you could have the JRSS low pass filter use the low pass filter setting in Room Correction. If it is too difficult, just leave it at 48 dB/octave regardless of what the user wants.
This makes Room Correction the only place to set a crossover, etc. It is just a few extra lines of code, right. ;D
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Last night I watched Valkyrie in a home theater with extensive room treatments. The treatments make a great difference (I heard the room before the treatments, too). This weekend I plan to build some bass traps in the front corners of my living room. I also want to do some more calibration after they are installed. I was reading about the DIY Audio Test DVD (http://www.avsforum.com/avs-vb/showthread.php?t=742969) at AVSForum and came upon the post I quoted below (post #291). Does MC's energy preservation calculations of moved bass arrive at similar numbers to the incoherent sound summing figures below?
OK, I keep getting asked so here it goes.
To set your system up for the recommended DD theatrical range you just follow the prompts on the DVD. Forget anything you read anywhere else about compensation, 75dBSPL, and anything else.
This will give you a max of 115dB from the LFE at 0dBFS, and 105dB from all the other speakers individually.
DTS is not recorded "HOT," neither are any DVDs. Most DVDs are dropped by -4dB. The ones that seem "Hotter" are just not as limited. There is no limiting on DTS (unless your processor does something funny.) Either way it is not "Hot," it just isn't attenuated by default.
Yes, some AVRs say +2 or -3 for DialNorm. These are setup where 0 on the screen(or no DialNorm screen in some cases) is actually -4dB below 0dBFS. In this case +4 is actually 0dBFS, and full loudness(DNR.) If you notice my DVD will pull up as +4. This is because there is no signal attenuation with my disk.
If you turn on Bass Management of any kind, and set at least one speaker to small you could get peaks above 115dB from your subwoofer. This is where it gets complicated.
If the sounds are not exactly the same this is called incoherent sound. There is a formula to figure it out but lets skip it. Lets assume this are 0dBFS level sounds;
LFE alone=115dBSPL
LFE+1Sp=115.41
LFE+2Sp=115.78
LFE+3Sp=116.12
LFE+4Sp=116.44
LFE+5Sp=116.74
LFE+6Sp=117.02
LFE+7Sp=117.28
If they are exactly the same they are coherent, and are a pain to sum, but here it is.
LFE alone=115dBSPL
LFE+1Sp=117.32
LFE+2Sp=119.25
LFE+3Sp=120.78
LFE+4Sp=122.09
LFE+5Sp=123.23
LFE+6Sp=124.23
LFE+7Sp=125.13
Now before you get too worried this is unlikely to ever happen. Why would you have full signal bass on every channel at the same time? There is also one other hurdle to clear, that never seems to get talked about.
Most people seem to know that the LFE channel is boosted +10dB. This is why 0dBFS is set at 115dBSPL on the LFE channel. What is mostly unknown is that before the crossover section in the process the LFE is digitally lowered -5dBFS, and is then boosted up to +15dB in an analogue output section to the RCA. All Bass Managed sound from the mains/surrounds are added in after the -5dBFS LFE step, and they are lowered by -15dBSF before being summed in with the LFE signal. (It is supposed to be boosted ~+15dB, but it isn't always.)
What this means is that if a sound on the Front Right speaker was supposed to play back at 105dB, it will still be output at 105dB. The thing that is missed is that there is only enough digital headroom left on the LFE channel for +5dB. Anything over that will clip the DSP. This means the max signal from the LFE should be 120dBSPL. Anything else is clipped off. The mastering process should have fixed any gross offenders, so around +5dB plus an extra dB should be it. See below for the extra dB.
There is the problem of Digital to Analogue conversion where the DAC can create a signal higher in dBV than the dBFS. This is known as 0dBFS+. If this perfect storm of chance were to happen you could end up with a peak higher than 120dBSPL. This shouldn't happen with commercially produced products, but it could happen. (Sound guys are human too, and forget the lessons from the analogue world still carry over.) It would be an abnormal situation though.
For more info:
Dolby Section 3.5:http://www.dolby.com/uploadedFiles/z...Guidelines.pdf
0dBFS+:http://www.tcelectronic.com/media/Le...per_AES109.pdf
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Last night I watched Valkyrie in a home theater with extensive room treatments. The treatments make a great difference (I heard the room before the treatments, too). This weekend I plan to build some bass traps in the front corners of my living room. I also want to do some more calibration after they are installed. I was reading about the DIY Audio Test DVD (http://www.avsforum.com/avs-vb/showthread.php?t=742969) at AVSForum and came upon the post I quoted below (post #291). Does MC's energy preservation calculations of moved bass arrive at similar numbers to the incoherent sound summing figures below?
I'm not exactly sure the question, but soho54's math makes perfect sense.
This was actually a helpful post because it made me realize it would be best to work the math so that you calculate the correct decibel level first, and work backwards to channel levels. We were working in the other direction.
So there will be some refinements to redirection energy preservation in a coming build.
Thanks.
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Let me expand. My question was that if you started with the LFE at 115db and the other channels at 105dB, would it sum to the same totals (based on number of channels) that soho54 got using the calculations that MC uses? If I am playing back 7.1 in MC and everything is at full signal strength (ie 115 & 105 dB), then is the combined LFE + redirected bass output going to equal around 117.28 dB? This is assuming the bass is slightly different in each channel
I'm glad the post was helpful. Now that the extensive bass management options are available in the public build, I'm thinking about starting a post at AVSForum's HTPC subforum.
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Let me expand. My question was that if you started with the LFE at 115db and the other channels at 105dB, would it sum to the same totals (based on number of channels) that soho54 got using the calculations that MC uses? If I am playing back 7.1 in MC and everything is at full signal strength (ie 115 & 105 dB), then is the combined LFE + redirected bass output going to equal around 117.28 dB? This is assuming the bass is slightly different in each channel
With build 15.0.92 or newer, the levels will match exactly.
Of course, there may be attenuation to avoid possible clipping, so instead of the LFE getting a 2.28 dB boost, the other channels get a 2.28 dB cut.
But remember that the notion of using "115 dB" and "105 dB" are sort of arbitrary. The math would work out the same if you used 60 dB and 50 dB.
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Wow, that was quick!
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How does the Crossover setting for the Subwoofer work, does it:
1) Cut everthing under XXhz over to the 0.1 channel (eg remove from L/R)
2) Add everthing under XXhz to the 0.1 channel (eg keep on L/R as well)
I like the option you get in FFDSHOW to chose the behaviour (and I personally like keeping the base on the main channels as well - I think you get a better integrated bass timber over the front sound stage)
Thanks
Nathan
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How does the Crossover setting for the Subwoofer work, does it:
1) Cut everthing under XXhz over to the 0.1 channel (eg remove from L/R)
2) Add everthing under XXhz to the 0.1 channel (eg keep on L/R as well)
The subwoofer crossover you select in DSP Studio > Output Format is used when simulating a subwoofer channel for sources that have no LFE channel of their own. For example, it comes into play when mixing 2.0 to 5.1 with JRSS.
It makes no changes to other channels.
I like the option you get in FFDSHOW to chose the behaviour (and I personally like keeping the base on the main channels as well - I think you get a better integrated bass timber over the front sound stage)
JRiver offers quite a lot more than FFDShow here. The key is that to do bass redirection (i.e. moving bass from other channels to the subwoofer), you use the 'Room Correction' DSP. We allow crossover frequency per speaker, and also configurable slope for the high and low pass filters.
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There are currently two DSP's that have bass management settings: Output Format and Room Correction.
Output Format
Like Matt mentioned, this setting only comes into play when you don't already have any information in the subwoofer channel. This would be the case if you are upmixing 2.0 to 5.1. If you already have content, like with a 5.1 or 7.1 source, then this setting is ignored. It uses a 48 db/octave low pass filter for the crossover frequency. It only copies the bass, but it doesn't remove it from the other channels.
Room Correction
This is handles bass management in two situations: 1) when you want to remove bass from other channels for content upmixed from 2.0 to 5.1 or 7.1 and 2) when you want to perform bass management on 5.1 or 7.1 sources.
This gives the option of crossover frequency, high and low pass filter slopes (they aren't labeled, but the top one is high pass filter for each channel and the bottom one is the low pass filter for the subwoofer channel), and whether the bass is removed (really only useful for upmixed sources), moved, or copied.
The Conflict
First, people see bass management settings in Output Format and don't realize that the settings that actually affect 5.1 or 7.1 content are under Room Correction.
Second, if Room Correction is set for Move Bass, like I assume most will use for 5.1 or 7.1 content, then these settings won't work right for upmixed 2.0 content. I suggested a method of integrating both settings earlier in this thread.
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Here is what the Bass Management in Room Correction looks like:
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So much to play with ;D
Anyway I "think" I am correct in saying from reading the above that
1) 2.0 Music --> 5.1 with base being "copied" to LFE (say a 85Hz Crossover with a 48db/ocatve low pass filter) is all done in Output Format by setting JRSS with my 2.0 Music I can get Base
2) 5.1 is then left as is
3) I can ignore Room Correction
If so I'm happy with that (see pic).
I'm still using FFDSHOW for both 2.0 --> 5.1 mixing in and LFE'ing in Video Sources till the JR Audio Renderer supports (and hence I can replace Reclock):
- Slave Audio to Video Clock, and (to a lesser extent)
- Change refresh rate to match Video frame rate
Thanks
Nathan
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- Slave Audio to Video Clock, and (to a lesser extent)
Would you mind starting a thread on this and explaining (again)?
There's really no such thing as "slaving". There is one filter in charge of the clock, but it's normally run as a system clock (we experimented with running it based on the sound card clock and it just doesn't work).
The audio render has to manage drift between the audio and video. I suppose it could use a rate adjuster for the audio signal (we already have this in our DSP Studio) to change audio playback rates slightly on the fly to account for drift instead of issuing corrections that the video has to sync to.
But I would like to understand if this is a real world problem or a theoretical problem. I've never seen a stutter during video playback. Have you tried the JRiver audio renderer? Are you actually seeing stutters?
Thanks.
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Matt - I've attached the ReadMe doco from Reclock, and while it is old the first few pages describe the various issues faced with Video playback from a PC better than I could. I use the term "slave" poorly but it is really about dynamically adjusting the audio sampling to keep in track with the video playback (rather than the other way around and as happens in CE Players). Like all "video/audiophile" things it is real for "some", but I'm sure most are oblivious to it all (and certainly for NTSC users you probably think it is normal as you have been consuming 3:2 pulldown material for ever, just like for us PAL users we are used to the 4% pitch increase on film based material). It is a complex interaction between all the parts of the chain and sometimes it can not be solved. My Intel G33 @ 24hz would repeat a frame every 17sec (about 1 per 400 frames) regardless of what I did. Once I noticed it, I was gone....I'd sit there counting down to the next one (very off putting when trying to be absorbed in the content). In reality we have 48,000 to 192,000 audio samples per second to play with yet as low as 24 for the matching video samples, so any dropped or repeated video Frame is going to be much more noticeable than resampled audio.
The reason I bring it up again, is that JR is spending alot of time and effort to ramp up the pure Audio side of the house for audiophile consumers and I thought I'd raise the focus on the video side as well, where I think we have:
- Sinc Audio playback to Video Clock,
- Auto change refresh rate to match Video frame rate
- Support high quality Video Renderers (eg madVR has never worked through "support" was added)
- I'm sure there are others!
I'll ask others far more knowledgeable than me to chip in.
Thanks
Nathan
PS - I'll let you split off this and the previous post to a new thread as I can not.
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OK - I did some testing with an MKV I made of Shooter (good opening scenes with lots of panning that shows up any inserted or dropped video frames as a stutter).
1) JR Renderer @ 50hz (my std refresh rate being in a PAL country) = Crap, lots of telecine judder
2) JR Renderer @ 24hz = Pretty good but about every 45 Sec I'd see a stutter during pans (not noticeable on still shots)
3) Reclock @ 24hz = Not stutter
Lets look at what is happening (see the reclock pic attached).
- Movie is encoded at 23.976fps (original was film so it is 24hz but tweaked for you NTSC countries)
- The actual refresh rate of my Video Card is 24.002hz
- Difference is about 0.1% or it is One Frame out of sync with the Audio each 42 Seconds (confirms what I see)
While the iPod'ers may not notice, imagine the uproar if there was an audio stutter every 45seconds from your audiophile crowd! It is no different on the Video side, once you notice it the stutter really gets under you skin...
You can see what Reclock is doing. It has increased the audio sampling from 48000hz to 48048 (or about 0.1%) so it is now in time with my Video Card clock of 24.002hz (which is 0.1% faster than the video). Reclock with keep changing this sampling rate so that that the Video Renderer neither needs to add or drop a frame.
Now one frame every 45 sec is not a big thing for most but if MC is to be a audiophile/videophile product such correction will be a plus for some.
Matt - I know you have golden ears but are not sure if video stutter is real so I'd suggest as a test:
1) Use a decent size TV that support 60hz, 50hz and 24hz
2) Play a (blu ray rip) with a good panning section at 60hz, 50hz and 24hz using the JR Renderer and see how the video flows
3) Do the same but this time with reclock
Hopfully your golden eyes will now see the stutter in poorly matched video timing!
I see that there are two potential options in MC:
1) Duplicate all the audio resampling & refresh rate changing stuff that Reclock does into the JR Audio Renderer or
2) Change the JR Audio Renderer so it is a transform filter and can still connect to the Reclock renderer. This way we can still use all of the DSP stuff you have made in MC + the stuff reclock already does so well.
Thanks
Nathan