INTERACT FORUM
More => Old Versions => JRiver Media Center 19 for Windows => Topic started by: Matt on July 17, 2013, 12:05:40 pm
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MC19 makes several important improvements to audio analysis and volume leveling:
- Adoption of the R128 industry standard (http://tech.ebu.ch/loudness) to analyze the loudness and dynamic range of content
- Ability to analyze audio for video files, including surround sound
- Smarter Volume Leveling that automatically respects intentional between track levels when playing from an album
- Volume Leveling works together with Adaptive Volume's peak level normalization (http://yabb.jriver.com/interact/index.php?topic=82023.0)
- Peak level is reported in decibels, measured as an R128 compliant True Peak, and reported per channel
This is a big topic, so I'll try to revisit it and post more details about the changes and the motivation at a later date.
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If you are using the oversampling method of calculating peak level as specified by the EBU, it might be worth changing this to "True Peak Level" (measured in dBTP) - you don't want to sell yourself short. :)
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Will backward compatibility be maintained so we can use the previous audio analysis method for old files and the new method for new files?
If not I will likely turn this feature off because I do not want to reanalyze my entire library.
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I think the problem is that mixing R128 and ReplayGain is not going to give good results.
ReplayGain is not nearly as good as R128 at leveling volume to the point that you don't have to touch the volume control any more.
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While I will have to re-analyze my files once true peak level analysis has been added, I am wondering about how downmixing factors into this.
I have analyzed a number of videos now, from downloaded content (e.g. made for YouTube or other sites) to television shows and films.
I haven't had a proper look over all the data, but there seems to be a number of items where volume leveling indicates that a positive correction is required, but peak level is already -1dB or higher. (and that's likely to increase once true peak measurements are implemented)
For example, I have an episode of The Wire which has a volume level of 4.7 LU, but a peak level of -1.3 dB, so volume leveling plays it back at +1.3 dB.
Even if I use Internal Volume or Parametric EQ set to -6dB as a test, Audio Path still indicates that Volume Leveling is only making a +1.3 dB correction. (P.S. when using the default Noire theme, Audio Path is not accessible during video playback in fullscreen mode for some reason)
The EBU R128 papers mention that separate analysis should be performed after downmixing, but you seemed to think that it wouldn't matter.
Would analysis after downmixing change things, or would the end result be the same?
It's still going to be very useful to have volume leveling active for downloaded videos, because they are often poorly mastered and the files I have analyzed so far range from -15.9 LU all the way to +22.0 LU, but it seems that it may not be very effective for commercially produced content - at least not if the target is -23 LUFS and peak level should not exceed -1 dBTP.
I seem to recall something being mentioned about possibly using -31 LUFS for some content in one of the EBU papers, but at some point I do wonder if we're just throwing away too much dynamic range in order to level things out though. (and when talking about losing 5-bits, it makes me wish my DAC accepted a 32-bit input…)
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A few more thoughts on analysis and volume leveling:
1. It would be great if analysis could be made to work smarter. Because analysis is now likely to be I/O bound (though maybe not as much, now that True Peak Level analysis is in) if there are files from multiple drives in the list, it should try to assign data from different drives to each CPU core.
2. Analysis should be prioritized by size, length, and bit-depth/sample rate, so that it gets through the maximum number of files in the minimum amount of time. I'm not sure which of these factors in the most when it comes to speed. (probably length, then sample rate?)
3. I think my system went to sleep last night when analyzing files. I had another process that was also keeping the system awake, and I think once that finished it went to sleep. (but that was only an hour or so before I went to use the computer)
4. With the multichannel Peak Level tags, it just returns [Varies] rather than an average for groups now.
5. Formatting in the Peak Level tag is still a bit difficult to read. Even without using a monospaced font, switching to monospaced characters (figure space, and figure dash) and changing the spacing a bit, in an attempt to keep each channel a similar length makes it a lot more legible:
(http://www.abload.de/img/monospacexzsqq.png)
The real solution would be to use a monospaced font, or align the channels some other way.
Note: I have found that a number of monospaced fonts do not support the figure space character, which is rather annoying. (but of course a regular space is now monospaced anyway)
6. Because analysis is now a lot more CPU intensive, there's a higher chance of it impacting playback, or other usage of the computer. If it's possible, moving this to a lower priority process would be really nice.
7. If the analysis window is currently open, selecting more tracks for analysis should add them to the bottom of the list.
8. Would it be possible to pause analysis, rather than being forced to stop it? Especially now that we are analyzing large files, there are times where I need to stop analysis temporarily because it's interfering with another operation on the computer. (e.g. I want to copy a file from a drive that is currently being analyzed)
9. When volume leveling is active, could internal volume or parametric EQ volume adjustments change the target level, rather than the post-gain level? If a file is not able to be fully normalized to prevent clipping, I'd like some way of increasing the headroom. I don't know that -23 LUFS is enough to properly level video on my system when I'm mixing stereo and downmixed 5.1/7.1 content.
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A few more thoughts on analysis and volume leveling
Just wanted to give a big thanks for your contribution mate, its being appreciated!
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I just tried to analyze a Blu-ray concert. The concert has two titles with a different part of the concert on each title. I have the main library entry and a Particle to handle the two titles. I first analyzed the main library entry. Then I analyzed the Particle. The Particle showed as already analyzed and the tags were filled even though it was a different title than the main library entry. I analyzed the Particle and the results were different so evidently both titles have been analyzed separately.
Also, the first time I ran Analyze Audio I had the stereo track selected. I changed it to DTS-HD and reran Analyze Audio. It considers it already analyzed and I had to uncheck "Skip analyzed files." I wonder if there is a way to keep analysis info for various audio tracks, or at least to reset the analysis state if a track has been switched.
Summary: Particles and different audio tracks show as having been analyzed if the either the main library entry or other audio track has been analyzed.
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I just tried to analyze a Blu-ray concert. The concert has two titles with a different part of the concert on each title. I have the main library entry and a Particle to handle the two titles. I first analyzed the main library entry. Then I analyzed the Particle. The Particle showed as already analyzed and the tags were filled even though it was a different title than the main library entry. I analyzed the Particle and the results were different so evidently both titles have been analyzed separately.
Also, the first time I ran Analyze Audio I had the stereo track selected. I changed it to DTS-HD and reran Analyze Audio. It considers it already analyzed and I had to uncheck "Skip analyzed files." I wonder if there is a way to keep analysis info for various audio tracks, or at least to reset the analysis state if a track has been switched.
Summary: Particles and different audio tracks show as having been analyzed if the either the main library entry or other audio track has been analyzed.
I don't know that there's a good or simple solution for this other than to either use the "best" track, or previously selected track if the file has already been played - which I think Media Center is already doing.
It's not an issue I have run into, or anticipate running into, because I just rip the best track and let JRSS handle the downmixing.
The problem is that if you're ripping the whole disc, that's potentially a lot of additional audio tracks you have to scan, if you want it to analyze everything. I have some films which have 5-10 audio tracks on them.
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Just like you mentioned with iso's, I just rip the whole movie with JRiver so I don't have to think about anything. Later I can go back with MakeMKV and cut out stuff I don't want. However, I recently realized that I hadn't set MakeMKV to use the dtsdecoderdll.dll file and was only decoding the DTS core. >:( By ripping the entire movie, there are no mistakes.
I wouldn't want it to analyze all the audio tracks either, but if I ever switch tracks because the default chosen track is the wrong one, I would want JRiver to reanalyze based on the current track. What if the Peak Level for the stereo track was -5.0 dB and the DTS-HD was +2.3 dB? If it had originally analyzed the stereo track before I switched tracks and won't reanalyze, there could be a problem.
It is probably no big deal, but just something to think about.
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I recently realized that I hadn't set MakeMKV to use the dtsdecoderdll.dll file and was only decoding the DTS core. >:( By ripping the entire movie, there are no mistakes.
I think this only matters if you are also converting to FLAC, but it's certainly easier to just rip to ISO.
I wouldn't want it to analyze all the audio tracks either, but if I ever switch tracks because the default chosen track is the wrong one, I would want JRiver to reanalyze based on the current track. What if the Peak Level for the stereo track was -5.0 dB and the DTS-HD was +2.3 dB? If it had originally analyzed the stereo track before I switched tracks and won't reanalyze, there could be a problem.
Well I don't think it should continue to use the same analysis, but I don't know that it should also force re-analysis. (because surely you're either playing the film, or finding the right track to rip if you're changing audio tracks)
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I think this only matters if you are also converting to FLAC, but it's certainly easier to just rip to ISO.
You are right. I originally converted to FLAC but haven't the past few years. I haven't used MakeMKV on a Blu-ray in a long time and forgot about that.
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I am pretty uneducated about this, but I have been using mp3gain to set levels for a number of years (instead of just MC) because mostly I play music on mp3players that aren't MC. I wanted the files modified for a standard player. Will MC19 be able to modify the files so they can play normalized outside of MC? I hope?
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I am pretty uneducated about this, but I have been using mp3gain to set levels for a number of years (instead of just MC) because mostly I play music on mp3players that aren't MC. I wanted the files modified for a standard player. Will MC19 be able to modify the files so they can play normalized outside of MC? I hope?
I think MP3 Gain is unique in that it will modify the gain of your MP3s in a non-destructive manner.
You're probably best to stick with MP3 Gain and use MC analysis when playing inside Media Center.
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I am pretty uneducated about this, but I have been using mp3gain to set levels for a number of years (instead of just MC) because mostly I play music on mp3players that aren't MC. I wanted the files modified for a standard player. Will MC19 be able to modify the files so they can play normalized outside of MC? I hope?
I think MP3 Gain is unique in that it will modify the gain of your MP3s in a non-destructive manner.
You're probably best to stick with MP3 Gain and use MC analysis when playing inside Media Center.
Thanks for asking and answering this question as I was doing/wondering the exact same thing.
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I am also in the same boat as Jack. I do MP3Gain first (for playing files outside my computer), then analyze from within JRiver. I have long thought that JRiver should add an MP3Gain-like modification during audio analysis. It would save me a lot of effort.
Though I am pretty uneducated about this as well, it would be great if JRMC19 would do this.
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I am also in the same boat as Jack. I do MP3Gain first (for playing files outside my computer), then analyze from within JRiver. I have long thought that JRiver should add an MP3Gain-like modification during audio analysis. It would save me a lot of effort.
Though I am pretty uneducated about this as well, it would be great if JRMC19 would do this.
One more guy who does it just like Jack, BigCat etc. I always use MP3Gain first and then MC. Would be great if I could throw MP3Gain away and use just MC for this purpose. Remember, like Jack said: Will MC19 be able to modify the files so they can play normalized outside of MC?
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One more guy who does it just like Jack, BigCat etc. I always use MP3Gain first and then MC. Would be great if I could throw MP3Gain away and use just MC for this purpose. Remember, like Jack said: Will MC19 be able to modify the files so they can play normalized outside of MC?
+1
see http://yabb.jriver.com/interact/index.php?topic=81626.0 (http://yabb.jriver.com/interact/index.php?topic=81626.0)
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Can I get some clarification on the Dynamic Range (R128) measurement? I'm trying to understand how this measurement correlates with the dynamic range of the track and the DR value obtained from the Dynamic Range Meter/Dynamic Range Database.
Fore example, Muse's "The Second Law" is a horribly compressed CD:
Track DR Dynamic Range (R128)
1 DR5 7.0 LU
2 DR5 7.4 LU
3 DR6 3.6 LU
4 DR7 11.8 LU
5 DR5 14.2 LU
6 DR5 12.4 LU
7 DR5 5.2 LU
8 DR6 11.3 LU
9 DR6 7.7 LU
10 DR6 8.4 LU
11 DR4 2.3 LU
12 DR6 8.1 LU
13 DR5 13.6 LU
And The Cure's "Kiss Me Kiss Me Kiss Me" which is nice and dynamic:
Track DR Dynamic Range (R128)
1 DR11 8.9 LU
2 DR12 2.7 LU
3 DR12 2.9 LU
4 DR13 5.2 LU
5 DR12 2.6 LU
6 DR13 3.4 LU
7 DR13 3.5 LU
8 DR12 3.0 LU
9 DR11 3.9 LU
10 DR14 3.0 LU
11 DR12 3.0 LU
12 DR12 6.4 LU
13 DR12 2.5 LU
14 DR12 3.5 LU
15 DR12 6.5 LU
16 DR12 3.1 LU
17 DR12 4.5 LU
The R128 is all over the place, tells me nothing about what I might hear and almost seems inverted. Lower LU means more dynamic range? The DR values are much more consistent with what I hear.
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A higher LU value should indicate higher dynamic range.
Media Center should be following the R128 spec: http://tech.ebu.ch/docs/tech/tech3342.pdf (http://tech.ebu.ch/docs/tech/tech3342.pdf)
I've just analyzed the EBU test files, and it seems to be working correctly.
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I think R128 has nothing to do with Dynamic Range. Why did you name it Dynamic Range?
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I think R128 has nothing to do with Dynamic Range. Why did you namend it Dynamic Range?
http://tech.ebu.ch/docs/tech/tech3342.pdf
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http://tech.ebu.ch/docs/tech/tech3342.pdf
http://tech.ebu.ch/docs/tech/tech3342.pdf --> Loudness Range should not be confused with other measures of dynamic range or crest factor
This is what I mean!
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The R128 is all over the place, tells me nothing about what I might hear and almost seems inverted. Lower LU means more dynamic range? The DR values are much more consistent with what I hear.
The TT Dynamic Range values are a measure of the crest factor of the music. Crest factor is the difference between average and peak levels in a track. This is based just as much on music style as whether there is compression used during mastering.
Actual dynamic range is the difference between the loudest and quietest parts of a track. The R128 values are a measure of the actual dynamic range of a track.
You can see from the Dynamic Range (R128) numbers from your two examples that "The 2nd Law" actually has more difference in dynamics than "Kiss Me Kiss Me Kiss Me." However, "The 2nd Law's" average levels are closer to its peak output which result in a much lower TT DR number.
This is what I mean!
The R128 Loudness Range is a measure of the difference between loudest and quietest portions of an audio track and should not be confused with TT Dynamic Range Meter which measures crest factor or the use of "dynamic range" to refer to the signal to noise content on a recording. JRiver is using Dynamic Range (R128) here because it probably has more meaning to people than Loudness Range.
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Hmm...I guess I was hoping to get a meaningful TT dynamic range like measurement. I don't see where MC19 returns the average R128 level. I'm assuming that would be -23 LU -(Volume Level(R128))? It would seem *in general* that a more negative Volume Level (R128) would indicate a more compressed/clipped file as long as the peak values don't drop as well. "The 2nd Law" averages about -15 LU for the Volume Level (128) with Peak Level (128) +0.5 dB while "Kiss Me Kiss Me Kiss Me" averages about -7LU for the Volume Level (128) and -0.5 dB Peak Level (R128). That would give "Kiss Me Kiss Me Kiss Me" about 7 LU more dynamic range which would correlate with the DR Database? Please correct me if I'm going off the deep end.
I do like the new volume leveling system.
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Hmm...I guess I was hoping to get a meaningful TT dynamic range like measurement.
Next build of MC19:
NEW: Added DR dynamic range analysis to the audio analyzer.
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Next build of MC19:
NEW: Added DR dynamic range analysis to the audio analyzer.
Woo hoo!
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Quick question, is audio analysis pretty much how it's going to be going forward? Are we going to need to reanalyze our library again after this point? I only ask because MC has been analyzing my library now for 3.5hrs and it's only about 1/6th of the way finished. So I'm looking at about 21hrs total. Would hate to have to do it again.
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Quick question, is audio analysis pretty much how it's going to be going forward? Are we going to need to reanalyze our library again after this point? I only ask because MC has been analyzing my library now for 3.5hrs and it's only about 1/6th of the way finished. So I'm looking at about 21hrs total. Would hate to have to do it again.
Since DR is coming next build (see above), you might wait to analyze.
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Next build of MC19:NEW: Added DR dynamic range analysis to the audio analyzer.
Awesome!
VP
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Next build of MC19:
NEW: Added DR dynamic range analysis to the audio analyzer.
Very Nice! You guys seem to get to nearly all my requests. Some quick, some not, but definitely always a great effort on your part.
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This made the upgrade completely worth it. Thanks!
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Since DR is coming next build (see above), you might wait to analyze.
Currently reanalyzing my whole library, I'm super excited to try all this out.
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Does anyone foresee any changes to audio analysis that may require a future reanalysis?
I have some 300,000 media files in my library and would like to save CPU cycles.
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Since DR is coming next build (see above), you might wait to analyze.
Currently reanalyzing my whole library, I'm super excited to try all this out.
Yep, I hear ya. I've got a separate machine set up just to do this one job. Audio only files didn't take too long. Analysing fine video's the "fun" job... ;)
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Currently reanalyzing my whole library, I'm super excited to try all this out.
Yep, I hear ya. I've got a separate machine set up just to do this one job. Audio only files didn't take too long. Analysing fine video's the "fun" job... ;)
Just finished up my re-analyzing last night, and I have to say, I'm pretty impressed so far. I cued up a play doctor playlist and listened for over an hour without having to adjust the volume once, which was never really the case with the old volume leveling (it would get close, but I'd find myself adjusting now and then even so). It's also neat to see all the new info about tracks. All in all a really neat improvement.
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Just finished up my re-analyzing last night, and I have to say, I'm pretty impressed so far. I cued up a play doctor playlist and listened for over an hour without having to adjust the volume once, which was never really the case with the old volume leveling (it would get close, but I'd find myself adjusting now and then even so). It's also neat to see all the new info about tracks. All in all a really neat improvement.
I agree - it's working really well when mixing random stereo tracks together.
When you introduce multichannel tracks - especially if you are downmixing to stereo - I think it could still use some work.
And album-based leveling now uses the gain from the loudest track rather than the average gain, which breaks leveling and has me changing the volume between albums again.
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In the Past I never saw the value in this feature... Now that I've been listening to the r128 version.. Its a real treat not always playing with volume. It works great !
Congrats JRiver. an thanks for the feature I thought I didn't want :P
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I agree - it's working really well when mixing random stereo tracks together.
When you introduce multichannel tracks - especially if you are downmixing to stereo - I think it could still use some work.
And album-based leveling now uses the gain from the loudest track rather than the average gain, which breaks leveling and has me changing the volume between albums again.
I noticed the change in the way albums were handled this morning. I guess the idea is that it's designed to automatically prevent clipping if you use it with adaptive volume? Not sure why else it would change.
I think I agree with you though, while it works great for playlists, it does seem result in some pretty serious inter-album differences. I just tried a "torture test": switching from Slayer (loudest track requires -16.5 LU) to Berlioz (loudest track requires -2.3 LU) (a 14.2 LU spread in the volume leveling). To get those sounding about "the same" I had to make about a 3.5 dB adjustment, after volume leveling, to get subjectively similar sound (total of 17.7 difference). The averages for those albums would have been around -16 LU and +2.2 LU respectively (an 18.2 LU spread).
In this admittedly very small sample, the average would have covered my perceived difference in volume within a half dB, while the new "pick the lowest" method is much farther off. I'm sure it won't work out that way in every case, but I'd be curious to hear additional observations on the new album leveling method.
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I noticed the change in the way albums were handled this morning. I guess the idea is that it's designed to automatically prevent clipping if you use it with adaptive volume? Not sure why else it would change.
Yes, I believe the change was made to avoid potentially clipping with album playback. But with analysis measuring the peak levels, it seems like it shouldn't be necessary.
I think I agree with you though, while it works great for playlists, it does seem result in some pretty serious inter-album differences. I just tried a "torture test": switching from Slayer (loudest track requires -16.5 LU) to Berlioz (loudest track requires -2.3 LU) (a 14.2 LU spread in the volume leveling). To get those sounding about "the same" I had to make about a 3.5 dB adjustment, after volume leveling, to get subjectively similar sound (total of 17.7 difference). The averages for those albums would have been around -16 LU and +2.2 LU respectively (an 18.2 LU spread).
In this admittedly very small sample, the average would have covered my perceived difference in volume within a half dB, while the new "pick the lowest" method is much farther off. I'm sure it won't work out that way in every case, but I'd be curious to hear additional observations on the new album leveling method.
This has been my experience too - there's a noticeable difference between playlists of albums now that it's using the loudest track's gain rather than the average.
I did some analysis on my library a couple of weeks ago, and about a third of my albums had a 3dB or greater delta, and 10% had a 6dB or greater delta when using the loudest track compared to the average level.
That's not to say that using the average level will be perfect either, but perceptually it seemed to give much better results.
I'm not convinced that clipping is going to be a widespread issue, because most music seems to be fine when normalized to -23 LUFS.
I do think we need some way to control the amount of headroom that volume leveling has available though; 23dB is not enough for video leveling - it needs more like 30dB.
Linking the internal volume control to the target volume level, rather than applying volume after leveling has been performed seems like the neatest solution for this. (if you want -30 LUFS as a target, reduce the volume by 7dB)
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Hello folks, I am new to this volume levelling (R128) feature.
I am confused.
Is it available to me now - if so how and where do I find it.
How do I use it; for example...what settings.
Is it still in beta and therefore subject to change.
thanks,
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Hello folks, I am new to this volume levelling (R128) feature.
I am confused.
Is it available to me now - if so how and where do I find it.
How do I use it; for example...what settings.
Is it still in beta and therefore subject to change.
1. You need to analyze (or re-analyze) your files in MC19. Right-click your files and select Library Tools > Analyze Audio. This might take a while.
2. Enable Volume Leveling in DSP Studio
There have been a number of changes thus far, so we don't really know if there will be any more. It seems like it's in a state where we aren't going to see any more changes, but you may potentially have to re-analyze your library by the time MC19 is out of beta.
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I'm not convinced that clipping is going to be a widespread issue, because most music seems to be fine when normalized to -23 LUFS.
Do remember that -23 LUFS is really designed for "broadcast" and keeping things under control in that environment. In my research on EBU 128 - I have seen several documents and studies that are promoting a movement of somewhere around -15 LUFS - being a more reasonable ballpark for music.
When I get MC 19 into my environment - I will be able to test this further....but in several previous tests with different software - i found -23 LUFS on a typical "non brickwalled" rock album to be excessive (to me) in driving down the levels too far.
YMMV.
Cheers!
VP
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I'm not convinced that clipping is going to be a widespread issue, because most music seems to be fine when normalized to -23 LUFS.
I do think we need some way to control the amount of headroom that volume leveling has available though; 23dB is not enough for video leveling - it needs more like 30dB.
Linking the internal volume control to the target volume level, rather than applying volume after leveling has been performed seems like the neatest solution for this. (if you want -30 LUFS as a target, reduce the volume by 7dB)
I had a question about that point, if you wouldn't mind expanding on it a little. My understanding is that internal volume happens "first." I say that because most of the DSP modules have an option for "process independently of internal volume" that undoes internal volume for them (which only makes sense if internal volume happens first). So what would the advantage be (from a clipping perspective) of linking internal volume to volume leveling?
Maybe an adjustable target for volume leveling might be a more straightforward solution, although I can understand if the devs are reluctant to move off of the studio standard.
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Do remember that -23 LUFS is really designed for "broadcast" and keeping things under control in that environment. In my research on EBU 128 - I have seen several documents and studies that are promoting a movement of somewhere around -15 LUFS - being a more reasonable ballpark for music.
When I get MC 19 into my environment - I will be able to test this further....but in several previous tests with different software - i found -23 LUFS on a typical "non brickwalled" rock album to be excessive (to me) in driving down the levels too far.
"Broadcast" includes radio - in fact that's where R128 has had the most adoption so far, I believe. Looking through the analyzed files in my library, 60% (of 15,000 files) require more headroom than -15 LUFS provides.
I had a question about that point, if you wouldn't mind expanding on it a little. My understanding is that internal volume happens "first." I say that because most of the DSP modules have an option for "process independently of internal volume" that undoes internal volume for them (which only makes sense if internal volume happens first). So what would the advantage be (from a clipping perspective) of linking internal volume to volume leveling?
I thought it was generally recommended to not use the "process independently of independent volume" option for anything that is processing the audio, and it's just for visualizers/analyzers.
The issue is that, because volume leveling happens first, if it's going to run into clipping, it does not level fully.
For example, say a track has a peak level of -1dBTP, but would require a correction of +6dB to be leveled properly.
Volume Leveling will play this back at 0dB, because it cannot raise the volume any higher than that, so it's now playing at -29 LUFS.
If you reduce the volume control by 6dB, the track is now playing 12 dB lower than the target level (-35 LUFS) because the volume adjustment happens after leveling.
If Media Center's internal volume control adjusted the target level, reducing the volume control would give you additional headroom:
With the volume control at 100%, the track will still play back at 0dB, which is 6dB quieter than the target level.
If you reduce volume to 88% (-6dB) and that adjusts the target to -29 LUFS, it will still play back at 0dB, but other tracks will be up to 6dB lower, because the adjustment gave you an extra 6dB of headroom.
If you reduced the volume control to 76% (-12dB; -35 LUFS) then that track would be played back at -6dB. (because it's supposed to be +6dB from the leveling target)
Maybe an adjustable target for volume leveling might be a more straightforward solution, although I can understand if the devs are reluctant to move off of the studio standard.
Well the reason I suggest that it be integrated with the volume control is that it adjusts the target level without introducing another control. I don't see why you would want to use a "normal" volume control when leveling is enabled.
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"Broadcast" includes radio - in fact that's where R128 has had the most adoption so far,
Haven't heard any radio stations here (Canada) that sound like they are using it - but I guess I should have specifically said television when I said "broadcast". With all the new "loudness" rules in effect here now - TV is where I notice LOTS of leveling has now occurred.
Radio has too many compressors and other doodads to make R128 worth the effort. And I do not thinking anyone is complaining about ads or music on the radio being too loud but I know (since I have actually called to complain) that ads on TV were out of control. I expect R128 plays a big role in that transition since it has become law now.
Looking through the analyzed files in my library, 60% (of 15,000 files) require more headroom than -15 LUFS provides.
I do not understand this - Can you explain how headroom a factor in applying an LUFS value?
Cheers,
VP
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"Broadcast" includes radio - in fact that's where R128 has had the most adoption so far, I believe. Looking through the analyzed files in my library, 60% (of 15,000 files) require more headroom than -15 LUFS provides.
I thought it was generally recommended to not use the "process independently of independent volume" option for anything that is processing the audio, and it's just for visualizers/analyzers.
The issue is that, because volume leveling happens first, if it's going to run into clipping, it does not level fully.
I think you might have it backwards (but I'm open to correction). You're right that it is recommended not to use "process independently of internal volume," but in the recommended state (off), internal volume is happening before DSP modules. Turning "process independently" on undoes internal volume for that module, which means that internal volume would otherwise happen first. That's why setting the internal volume lower gives you more headroom for DSP that involve boost, like EQ or convolution. Matt, I think, has confirmed that internal volume gives you more DSP headroom in the PEQ and convolution contexts. Otherwise it would be impossible to use any DSP that involved boost without constant risk of engaging clip protection, right?
If Media Center's internal volume control adjusted the target level, reducing the volume control would give you additional headroom:
With the volume control at 100%, the track will still play back at 0dB, which is 6dB quieter than the target level.
If you reduce volume to 88% (-6dB) and that adjusts the target to -29 LUFS, it will still play back at 0dB, but other tracks will be up to 6dB lower, because the adjustment gave you an extra 6dB of headroom.
If you reduced the volume control to 76% (-12dB; -35 LUFS) then that track would be played back at -6dB. (because it's supposed to be +6dB from the leveling target)
My understanding is that internal volume already gives you additional headroom, at least with other DSP modules (but I believe for volume leveling too). For example, I have +5 dB shelf that, if it were processed before internal volume, would cause clipping all the time. So I set my maximum internal volume -6 dBFS, with the result that I've never observed clipping in the analyzer even with very loud material. It's possible that PEQ and Volume leveling are different in this respect?
Thanks for the additional explanation, I'm a little dense sometimes ;D I'd be curious if one of the devs could confirm one or the other.
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I do not understand this - Can you explain how headroom a factor in applying an LUFS value?
1. R128 Peak level cannot be higher than -1.0 dBTP to avoid inter-sample clipping. (this is to account for some variance in the inter-sample peak calculation)
2. -23 LUFS is the target volume for leveling.
Using a real world example, you might have a track which has a [Peak Level (R128)] of -0.8 dBTP, and a [Volume Level (R128)] of 4.0 LU.
This means that the track needs to be played back at +4dB to sound like it's the same volume as other tracks.
Because clipping protection does not allow the peak level to exceed -1.0 dBTP, the track actually gets played back at -0.2 dB.
So to level this track properly, you need an additional 4.2dB of headroom; the target for leveling needs to be reduced to -27.2 LUFS, rather than -23 LUFS, for it to play back at the same perceived level as other tracks - either that or you just accept that this track is 4.2dB quieter than everything else.
If the target level was changed from -23 LUFS to -15 LUFS, this particular track would now be 12.2 dB quieter than any other track which is able to be normalized without clipping.
I think you might have it backwards (but I'm open to correction). You're right that it is recommended not to use "process independently of internal volume," but in the recommended state (off), internal volume is happening before DSP modules.
Volume Leveling seems to happen first. This is the track I mentioned above, which is supposed to play at +4.0 dB:
(http://www.abload.de/img/volumeeyudp.png)
Turning "process independently" on undoes internal volume for that module, which means that internal volume would otherwise happen first.
It's not an option for Volume Leveling.
EDIT:
In my library, I actually have a track (classical music) that needs an additional 7dB of headroom to be leveled properly, which would require the target be reduced to -30 LUFS!
But overall, -23 LUFS seems like a good target, as only 1% of my library (151 tracks) require more headroom than it provides.
This expression will let you see how much headroom your tracks require. (it could probably be cleaned up - but it works)
Delimit(if(isempty([Peak Level (R128)]),,formatnumber(math(removecharacters(left([Peak Level (R128)],5),/ /+,0)+RemoveCharacters([Volume Level (R128)],/ LU,0)+1),1)),/ dB,)
So if you were thinking about changing the target level to -15 LUFS rather than the current -23 LUFS, you would count all tracks that are -8.0 dB or greater. (note: positive values sort to the end of the list for some reason)
While -23 LUFS seems like a good target for music, video seems like it will require at least -30 LUFS to level properly, and we really need some way to analyze downmixed audio if multichannel files are being played back in stereo, for proper leveling.
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Because clipping protection does not allow the peak level to exceed -1.0 dBTP, the track actually gets played back at -0.2 dB.
So to level this track properly, you need an additional 4.2dB of headroom; the target for leveling needs to be reduced to -27.2 LUFS, rather than -23 LUFS, for it to play back at the same perceived level as other tracks - either that or you just accept that this track is 4.2dB quieter than everything else.
If the target level was changed from -23 LUFS to -15 LUFS, this particular track would now be 12.2 dB quieter than any other track which is able to be normalized without clipping.
Volume Leveling seems to happen first. This is the track I mentioned above, which is supposed to play at +4.0 dB:
Oh that ties that up; you're right (I was just looking at DSP studio myself and noticed what you describe). Internal volume happens before almost everything else, but not volume leveling. I have large number of tracks/albums that need positive volume leveling adjustments but have near 0 dBFS peaks, and it sounds like they're unlikely to get the correct adjustments unless internal volume is applied first. The vast majority of my listening happens at around 20% internal volume, so I'd have all the headroom I could ever need under those circs.
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Using a real world example, you might have a track which has a [Peak Level (R128)] of -0.8 dBTP, and a [Volume Level (R128)] of 4.0 LU.
Do you have very many real world examples in your library where the Peak Level (R128) doesn't have enough headroom to adjust for Volume Level (R128)? In my library of 6400 audio tracks I found just three and it only made a 1.6 dB or less difference. I also would never play back those tracks without the entire album so it wouldn't matter anyway. The one needing the most Volume Level (R128) measures 15.8 and it has a Peak Level (R128) of -15.1 dBTP. The actual adjustment is 14.1 dB.
Only 200 tracks of mine need a positive volume adjustment (.03%). Out of those 200, there are only 4 tracks that I would play in a mixed playlist.
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Do you have very many real world examples in your library where the Peak Level (R128) doesn't have enough headroom to adjust for Volume Level (R128)? In my library of 6400 audio tracks I found just three and it only made a 1.6 dB or less difference. I also would never play back those tracks without the entire album so it wouldn't matter anyway. The one needing the most Volume Level (R128) measures 15.8 and it has a Peak Level (R128) of -15.1 dBTP. The actual adjustment is 14.1 dB.
As I mentioned before, only 1% of the music I have analyzed so far (151/~16,000) requires more headroom than -23 LUFS provides; and most of those tracks are classical music, which would generally be played back as albums anyway. I think -23 LUFS is a very good target level for music.
Vocalpoint suggested that the target be raised to -15 LUFS though - only 38% of my library would be able to be properly normalized using that as a target level.
I think if people are finding that playback is too quiet when using Volume Leveling, the solution is to use it in conjunction with Adaptive Volume set to Peak Level Normalize, rather than raise the target level. This will ensure the current playlist is level, but plays it as loud as possible while avoiding clipping.
For videos, however, more than 50% of my library requires more headroom than -23 LUFS provides. The worst offender is a film which requires 11.9dB of additional headroom.
Overall, it seems that -30 LUFS would be a suitable target for most films though. I'm not suggesting that the default be changed, but I would like some way of adjusting it. (and using Internal Volume seems like the neatest solution)
But that doesn't tell the whole story - when you downmix to stereo, the peak and average levels can change quite a bit. Some files require additional headroom, and others don't need nearly as much.
Matt has tried a few ways of estimating it, but perceptually, I don't think it works very well for films. It seems to do a better job with music, mostly being ±3dB with the files I have tested so far.
I think the only real solution is to have an option that will also analyze the downmix. There are reasons why it's not a good idea to do this, but I can't come up with a better solution.
But it could also be argued that volume leveling is not as much of a concern for film playback, because they're going to be at least 90 minutes long, and it's not as common to be playing them back-to-back.
It's mostly a problem for videos that have been created for the web (YouTube etc.) which are stereo, but simply need more headroom.
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Do you have very many real world examples in your library where the Peak Level (R128) doesn't have enough headroom to adjust for Volume Level (R128)? In my library of 6400 audio tracks I found just three and it only made a 1.6 dB or less difference. I also would never play back those tracks without the entire album so it wouldn't matter anyway. The one needing the most Volume Level (R128) measures 15.8 and it has a Peak Level (R128) of -15.1 dBTP. The actual adjustment is 14.1 dB.
Only 200 tracks of mine need a positive volume adjustment (.03%). Out of those 200, there are only 4 tracks that I would play in a mixed playlist.
For me, about 1500 out of 56000 tracks require a positive adjustment, and about 1/2 of those (a little less than 800) require a positive adjustment greater than the peak level will accommodate, so this affects a little less than 2% of tracks for me. The majority of those (about 2/3) are classical music, so probably not playlist fodder, but a substantial number of the remainder might wind up in playlists.
It's a small issue (for me), and I'm sure there are some unintended consequences I'm not seeing. But if it were frictionless, it seems like it would be preferable to process internal volume first or in composition with volume leveling so leveling would work for those tracks too.
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how do I go about displaying the DR value of an analyzed song?
thanks
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Vocalpoint suggested that the target be raised to -15 LUFS though - only 38% of my library would be able to be properly normalized using that as a target level.
I have since found out that ReplayGain is essentially = -18LUFS. And I finally found the document that I was thinking about in relation to my -15 LUFS suggestion...and it's here...
http://techblog.studio-compyfox.de/media/tech-doc/tech001_2012-Q2_K-System_v2.pdf
This document combines the best attributes of R128 with the Bob Katz "K System" and proposes a different target LUFS with the respect to music production. I feel this -16 LUFS standard is more in line with what one would have been experiencing with ReplayGain and it addresses the sharply quieter playback that the standard -23 LUFS target gives. That standard target was created to address the needs of ongoing broadcast material (voices, spots, music etc) in TV/radio and to me - simply is too aggressive a target to apply to my own library from home use.
I will be able to speak to this much better of course when I install v19 - but we are not ready for that just yet.. :)
I hope you will give the doc a read and let me know what you think.
Cheers!
VP
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I have since found out that ReplayGain is essentially = -18LUFS. And I finally found the document that I was thinking about in relation to my -15 LUFS suggestion...and it's here...
http://techblog.studio-compyfox.de/media/tech-doc/tech001_2012-Q2_K-System_v2.pdf
This document combines the best attributes of R128 with the Bob Katz "K System" and proposes a different target LUFS with the respect to music production. I feel this -16 LUFS standard is more in line with what one would have been experiencing with ReplayGain and it addresses the sharply quieter playback that the standard -23 LUFS target gives. That standard target was created to address the needs of ongoing broadcast material (voices, spots, music etc) in TV/radio and to me - simply is too aggressive a target to apply to my own library from home use.
I will be able to speak to this much better of course when I install v19 - but we are not ready for that just yet.. :)
I hope you will give the doc a read and let me know what you think.
Cheers!
VP
I seem to recall that JRiver's implementation of the ReplayGain standard was about 6 dB different than the actual standard (but tagged in a way that allowed compatibility with players using the normal standard). The specification for the official ReplayGain standard target is -14 LUFS (89dB calibration) http://wiki.hydrogenaudio.org/index.php?title=ReplayGain_specification#Clipping_prevention, and JRiver's implementation was -20 LUFS (83 dB calibration) to bring it in line with film mastering/movie theater calibration.
So the new volume leveling target will be about 3 dB quieter than the old volume leveling target (bracketing out other differences in the way the standards work that might affect how successfully they're hitting the target). Subjectively, for what it's worth, the new volume leveling doesn't sound much quieter to me than the old volume leveling, just much more consistent in it's effects.
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I seem to recall that JRiver's implementation of the ReplayGain standard was about 6 dB different than the actual standard (but tagged in a way that allowed compatibility with players using the normal standard). The specification for the official ReplayGain standard target is -14 LUFS (89dB calibration) http://wiki.hydrogenaudio.org/index.php?title=ReplayGain_specification#Clipping_prevention, and JRiver's implementation was -20 LUFS (83 dB calibration) to bring it in line with film mastering/movie theater calibration.
Yes - however I still added "+6db Fixed" in Volume Leveling to my playback - essentially pushing the JR implementation back to the RG standard. I have found this to be absolutely perfect for my entire library.
So the new volume leveling target will be about 3 dB quieter than the old volume leveling target (bracketing out other differences in the way the standards work that might affect how successfully they're hitting the target). Subjectively, for what it's worth, the new volume leveling doesn't sound much quieter to me than the old volume leveling, just much more consistent in it's effects.
Good to know. I look forward to using the new system...
Q: Within all the changes made here - is it possible to just dip my toe into the R128 pool slowly when transitioning to v19? That is to say - can I count on my existing ReplayGain values to work as usual in v19 or do I have to go all in with R128 right off the bat?
For me - my audio analysis metadata is critical and I certainly do not want to trash my library (or prematurely stop using RG) in one fell swoop when I make the move to v19. Would rather experiment (and potentially move away from RG) in a slow graduated process before going all in with R128.
Appreciate any comments on this..
VP
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how do I go about displaying the DR value of an analyzed song?
Right click a column and select Dynamic Range (DR) to add it.
I have since found out that ReplayGain is essentially = -18LUFS.
The ReplayGain 2 specification (http://wiki.hydrogenaudio.org/index.php?title=ReplayGain_2.0_specification#Reference_Level_and_Gain) is based upon the ITU BS.1770-2, which is what the R128 specification that JRiver have implemented in MC19 is based upon. It also specifies a target level of -23 LUFS. (actually denoted LKFS in the spec, but it's the same thing)
And I finally found the document that I was thinking about in relation to my -15 LUFS suggestion...and it's here...
http://techblog.studio-compyfox.de/media/tech-doc/tech001_2012-Q2_K-System_v2.pdf
This document combines the best attributes of R128 with the Bob Katz "K System" and proposes a different target LUFS with the respect to music production. I feel this -16 LUFS standard is more in line with what one would have been experiencing with ReplayGain and it addresses the sharply quieter playback that the standard -23 LUFS target gives. That standard target was created to address the needs of ongoing broadcast material (voices, spots, music etc) in TV/radio and to me - simply is too aggressive a target to apply to my own library from home use.
This paper is focused on music production rather than playback, and the argument throughout seems to be that there will be resistance to adopting -23 LUFS in production because it's not loud enough - which is exactly the kind of thinking that got us where we are now with highly compressed dynamics in the first place. (because it used to be that broadcast used peak normaliation) The proposed "K-System v2" is to make the transition to -23 LUFS easier, rather than going directly to it for production.
But the whole point of R128 is that it doesn't matter how loud a track was produced, it will be normalized to -23 LUFS regardless, and it puts tracks with highly compressed dynamics at a disadvantage. (because the compression was used to make them loud)
-16 LUFS means that one third of my library is unable to be normalized. Even if I remove classical tracks, it's still 27% of my library.
And in broadcast, some radio stations have already implemented R128 normalization.
I will be able to speak to this much better of course when I install v19 - but we are not ready for that just yet.. :)
I don't think it's fair to be criticizing the -23 LUFS target, if you haven't tried it. R128 normalization ≠ ReplayGain v1.
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I don't think it's fair to be criticizing the -23 LUFS target, if you haven't tried it. R128 normalization ≠ ReplayGain v1.
Well - I did not say (nor imply) that I have not tried/used R128.
I have been using it for a long while within my pro audio work. But normalizing a typical pop/rock track to the -23LUFS target within current tools like Wavelab 8 - which has a state of the art R128 conversion tool - is in my opinion - a perhaps a tad much. I guess I need to understand it more.
Now I did say I have not tried R128 with MC v19. Hence the reasons for my questions. Just trying to understand what's new and what's available.
Cheers,
VP
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Well - I did not say (nor imply) that I have not tried/used R128.
I have been using it for a long while within my pro audio work. But normalizing a typical pop/rock track to the -23LUFS target within current tools like Wavelab 8 - which has a state of the art R128 conversion tool - is in my opinion - a perhaps a tad much. I guess I need to understand it more.
Now I did say I have not tried R128 with MC v19. Hence the reasons for my questions. Just trying to understand what's new and what's available.
Ah I remember we had this discussion a while ago (http://yabb.jriver.com/interact/index.php?topic=79578.msg543243#msg543243).
Your editing tools do not appear to be simply normalizing the volume to conform to the R128 spec, but making other changes to the waveform such as adjusting the dynamic range.
R128 analysis and leveling only adjusts the volume level during playback - nothing else.
The point is that R128 analysis gives a much more accurate measurement of the track's perceived loudness than ReplayGain did, and now includes checks to prevent inter-sample clipping.
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Just want to give the MC Team KUDOS for developing this new volume leveling. As I use JRiver mainly as an audio player for my album library, I'm noticing an immediate improvement in audio quality and this alone was well worth the MC19 upgrade.
On the minus side, STILL no MiniView options,,,,,,,,,,, ?
Cheers,
Robin
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What is the difference between Peak Level (R128) and Peak Level (Sample)?
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What is the difference between Peak Level (R128) and Peak Level (Sample)?
Peak Level (sample) measures the peak level encoded inside the file. The maximum this can be with a lossless file, is 0dB. Lossy files may have samples that are above 0dB, as they use floating point values.
Peak Level (R128) upsamples the audio before measurement, and then measures the peak level. This is known as True Peak Level, and is measured in dBTP.
This is done because when you convert a digital signal to an analog waveform, it's possible to have signal levels that go above 0dB. Particularly with modern music releases, tracks are often adjusted so that the peak sample level is encoded at -0.1dB.
Most DACs do not seem to have headroom built into them to account for this, because any headroom you leave is effectively "throwing away" SNR (http://en.wikipedia.org/wiki/Signal-to-noise_ratio), so you have the possibility of clipping/distortion on playback. The only DAC I am aware of that has headroom built in and advertises this fact, is Benchmark's DAC2 models. (that doesn't mean it's the only DAC with headroom though)
This clipping is known as an inter-sample peak. The Peak Level (R128) measurement is not going to be 100% accurate (speed of analysis is a factor, and different DACs may produce different results) so there's an additional 1dB headroom to account for this variance.
Here's an example that I pulled off Google Images which illustrates the problem:
(http://www.abload.de/thumb/illegal_signalvwudn.jpg) (http://www.abload.de/img/illegal_signalvwudn.jpg)
In this image, you can see that none of the samples (the squares) exceed 0dB - but the analog waveform those samples represent exceeds 0dB. This would result in clipping/distortion when played through most DACs.
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Thanks, 6233638, for the explanation.
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Here is an article along the same lines:
http://www.indexcom.com/tech/0dBFS+/
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I'm still using MC17, but am thinking of upgrading when MC19 is available. But a lot of the technical talk is over my head.
1) Is this feature something I can turn on and off? I like to list to albums a lot, and I don't want it affecting volume between tracks
2) Does this feature negatively impact sound quality in any way? I use FLAC and try to get the most out of these files. I don't want something interfering or changing this.
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1) Is this feature something I can turn on and off? I like to list to albums a lot, and I don't want it affecting volume between tracks
Yes, you can turn it off. Albums are played back in such a way that all tracks are adjusted by the same amount.
2) Does this feature negatively impact sound quality in any way? I use FLAC and try to get the most out of these files. I don't want something interfering or changing this.
The short answer is that there should not be an audible change.
With a target level of -23 LUFS, you are theoretically discarding up to 4-bits of precision (6dB per bit) but with a 24-bit output that should still give you 120+dB of SNR which is beyond most DACs anyway, and still sufficient for "lossless" CD quality playback. (16-bit)
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For some reason I am noticing a degradation of sound quality when using volume levelling, a not too subtle loss of clarity to the sound. I thought I'd take a look in here to see if others had experienced the same, but it appears not. It's OK for shuffle play when music is just a background thing but when using headphones and listening critically I don't like it at all. For now this feature is staying off. It's not really a big deal, I don't mind adjusting the volume as I need to.
All my files are ripped as aiff at the same sample rates as the originals, so mostly 16/44.1 with some 24/96 or 192 downloads & even a couple of DsD's as my DAC is DsD capable. I never compress anything. Don't see why that would make any difference though. I do often use DsD encoding and quite like it.
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If you are outputting 16-bit files at 24-bit or greater (which Media Center does by default) you should not hear any degradation.
However, with the way that our hearing works, louder generally sounds better. So if you have enabled volume leveling and have not matched the volume to its previous level, you may perceive it to sound worse.
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Yes, you can turn it off. Albums are played back in such a way that all tracks are adjusted by the same amount.
The short answer is that there should not be an audible change.)
Some of this discussion is admittedly over my head.
For clarity, is this a non-destructive process? There are no actual changes to audio files?
Are values written as tags in the files themselves, or stored in the library?
I ask because I have been using dbpoweramp to rip and convert files and add RG tags in the process. There is a DSP for R128 Normalization, but it rewrites the audio content of the files so i have not used it.
Thanks.
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For clarity, is this a non-destructive process? There are no actual changes to audio files?
Are values written as tags in the files themselves, or stored in the library?
It is non-destructive. The audio in the file is not changed in any way.
Values are stored in the library and tags and optionally used at playback time to enhance the experience. You can turn the playback processing on or off easily.
I'm a believer in playback-time processing. Store a perfect copy, then apply any processing at playback time. This provides more flexibility and provides room for future improvements without ever requiring you to rerip / reencode.
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It is non-destructive. The audio in the file is not changed in any way.
Values are stored in the library and tags and optionally used at playback time to enhance the experience. You can turn the playback processing on or off easily.
I'm a believer in playback-time processing. Store a perfect copy, then apply any processing at playback time. This provides more flexibility and provides room for future improvements without ever requiring you to rerip / reencode.
Appreciate the clarification. I'm also a believer in playback processing. My music collection is large enough that re-ripping it would be a gigantic chore. I take pains to ensure I have multiple current back ups of my entire library, specifically to avoid having to rip it over again.
Does MC19 still support ReplayGain tags?
I've used RG solely because it was non-destructive. I'm hoping the dbpoweramp folks add a DSP in future that will analyze and store the R128 info as tags.
I'll upgrade to MC19 now, and give R128 a try.
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Does MC19 still support ReplayGain tags?
RG values are derrived from R128 values upon analysis with MC19. If there are no R128 values but there are previous RG values, MC falls back to using them.
At least, that how I understand how it works now.
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I installed the MC19 beside my MC18 in W8 and it worked as well as with MC18 as far as I can see and hear. (I only use MC18/19 for two channel stereo). Also Gizmo in my Android smartphone got it without special actions. Fine so far...
However when using "Play Now" from Gizmo showing images on the laptop plus titel and time behaves different now. When I start a new album, the cover is shown like before with readable title in negative White plus the option i use in under the gadgets meny. After some seconds the cover is fading out (like before) and the program import images from internet. Just as with MC18. But after 3-4 minutes the text is fading down in light and can't be read. When a new tune is loaded, the album cover normally shows up again but now it is so faded so it is hardly visable.
Is this something I can correct? If so, where? In MC18 it is not an issue.
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Is it possible to display the album DR value instead of single tracks DR value?
Thanks and bye
Andrea
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Is it possible to display the album DR value instead of single tracks DR value?
Thanks and bye
Andrea
http://yabb.jriver.com/interact/index.php?topic=82810.0
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thank You for the answer but what I'd like to have is a field/expression called "Album DR" like the one showed with dynamic range tool on foobar to have a view grouped by that value; for example I use a sample rate grouping (see attached file) and I'd like to do the same thing with Album DR value
Anyone knows if it's possible?
Thanks and bye
Andrea
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It should be possible to use the calculated album DR values from the thread I linked in a custom field, It shouldn't be too hard but I haven't had time to figure it out yet.
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Please provide an option to (re)analyze loudness only but keep existing BPM values. As MC sometimes comes up with completely irrelevant values, I always analyze and enter them myself, and I really don't want to lose all that effort I had put into it already...
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Please provide an option to (re)analyze loudness only but keep existing BPM values. As MC sometimes comes up with completely irrelevant values, I always analyze and enter them myself, and I really don't want to lose all that effort I had put into it already...
You could make your own field and copy the existing BPM values into it.
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You could make your own field and copy the existing BPM values into it.
Thanks for the tip although it's only a workaround.
Searching the history here I found out this has been already asked as a feature request in past. With regards to this response of yours do I get it right you simply don't want to add such an option? Why?
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I think because if they start with excluding BPM, someone else comes along asking for excluding DR values and then another wants to keep RG values. Before you know it, the analyser becomes cluttered with options to exclude each individual item. I, for one wouldn't like that.
Temporarily copying the BPM values to custom field is really quite simple and IMHO the best option. If you don't save the new field to files its very fast too.
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Need some direction on how to make sure I am setting this correctly. I have re-analyzed audio on my entire collection to the R128 std. While playing my library yesterday I had wild swings in vol. I should note I am streaming via DNLA to my Oppo BDP-105. Previously I could set this in audio options. Now it is grayed out saying I must adjust via Media Network/DNLA server but I see no option to do this. I also use JRemote which has no option to set this.
Any help to get this set up will be greatly appreciated
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Make sure you have the latest build of MC19. A DLNA bug was fixed that probably caused this.
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I updated to the latest. My question now is do I configure Volume Leveling/Adaptive Volume for the Player or the Device (Oppo)? Not avail for Device (Oppo). So if setting at the Player will apply VL to my media server, what are the best settings to make sure it is setup correctly?
Thank you in advance
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I don't like the new leveling system - yet anyway. It brings the volume down so low that it's impossible to play something that's been analyzed along side something that hasn't. Using Adaptive Volume helps but seems to effect the direct audio connection for some files. I think it's because I'm used to using Replay Gain with a fixed adjustment of +8db. I have paid for the version 19 upgrade but will continue to use 18.
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This is fixed in the next build:
1. Fixed: Adaptive volume would sometimes not apply (fixed) volume adjustment on already analyzed files.
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This is fixed in the next build:
1. Fixed: Adaptive volume would sometimes not apply (fixed) volume adjustment on already analyzed files.
Thanks Jim.
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Hello Jim,
I re-analyzed all my music library on MC 19....32 build, last weekend took me around 33hrs to do so, Huge library. Today I came and started MC19 and tried to check some values on the analyzed data, and it looks like everything is lost, all the library seems to be unanalized. What may happened?
John
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Jim
I am playing an album reanalyzed in MC19 and applying volume leveling and adaptive volume>peak level normalize and it clips terrible. Volume leveling is at -12.9 and peak normalization is +12.0 and is clipping really badly even having clip protection on. On MC 18 i didn't have this issue of course because I was applying +3 of fixed gain on volume leveling.
I hope this issues can be resolve soon, In the meantime I am back on MC18.
John
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I am playing an album reanalyzed in MC19 and applying volume leveling and adaptive volume>peak level normalize and it clips terrible. Volume leveling is at -12.9 and peak normalization is +12.0 and is clipping really badly even having clip protection on. On MC 18 i didn't have this issue of course because I was applying +3 of fixed gain on volume leveling.
I hope this issues can be resolve soon, In the meantime I am back on MC18.
If you have analyzed the files with MC19, it should not be allowing them to clip, because it measures the true peak level (the Peak R128 value) and does not allow it to go above -1.0 dB.
If you want the volume level, you need to use Volume Leveling enabled on its own. Adding a +3dB adjustment (which would be equivalent to +9dB now) will not allow most files to be levelled properly. R128 requires a target of -23 LUFS (dB) for this.
Volume Leveling combined with Adaptive Volume will level all tracks in the current playlist and then push it to the maximum volume it can, while avoiding clipping.
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6233638 thanks for your reply, but maybe something is wrong because it really clips and badly, and if I use only vol leveling it goes very quite, and I need to crank my amp the vol up a lot. And I found there are several albums doing this clipping.
Anyway I hope this things get resolved in a new build, In the mean time using MC18,
Thanks for the explanation
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6233638 thanks for your reply, but maybe something is wrong because it really clips and badly, and if I use only vol leveling it goes very quite, and I need to crank my amp the vol up a lot. And I found there are several albums doing this clipping.
Anyway I hope this things get resolved in a new build, In the mean time using MC18,
Thanks for the explanation
What are the Peak Level (R128) values for the files? Perhaps they need to be re-analyzed.
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If you have analyzed the files with MC19, it should not be allowing them to clip, because it measures the true peak level (the Peak R128 value) and does not allow it to go above -1.0 dB.
If you want the volume level, you need to use Volume Leveling enabled on its own. Adding a +3dB adjustment (which would be equivalent to +9dB now) will not allow most files to be levelled properly. R128 requires a target of -23 LUFS (dB) for this.
Volume Leveling combined with Adaptive Volume will level all tracks in the current playlist and then push it to the maximum volume it can, while avoiding clipping.
Do you have anything else enabled? Equalizer, PEQ, Room Correction?
I found that some of the other options are not considered by volume leveling and can cause clipping.
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What are the Peak Level (R128) values for the files? Perhaps they need to be re-analyzed.
+0.1 dbTP and also some are 0.0dbtp, i did the re-analyzed and is just the same. this weird. Also coming back from MC18 all my files appeared as not analized. this is frustrating it took me like 30 hours to analyze all track, 34K+ track. I don't know if MC19 will work for me.
Thanks
John
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you are right, if I combine EQ with effects it is when clips, if i do one or the other it is ok. This should be fixed in some way. because in EQ i really do a very slight change also on effects adding a little environment.
Thanks
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you are right, if I combine EQ with effects it is when clips, if i do one or the other it is ok. This should be fixed in some way. because in EQ i really do a very slight change also on effects adding a little environment.
Thanks
Volume leveling happens on the input signal. It is completely independent of later processing like EQ.
If you apply effects that could cause clipping, you have a few choices:
1) Switch to Internal Volume. This is my favorite solution, but I understand it only works well in some installations. Explanation here: http://wiki.jriver.com/index.php/Volume#Internal_Volume_Headroom
2) Add an effect to turn the volume down (equalizer preamp, parametric volume reduction, etc.). It doesn't matter where in the chain, since it's not a clip until output.
3) Just let the final safety valve detect the clip and turn the volume down. This is less of an audiophile solution, but the real world results will be pretty good.
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Probably a dumb question: the results of the analysis are stored in the library? So if, for instance, I have a need to restore/re-image my machine to a pre-analysis period, simply restoring a saved backup would have all the analysis data? No need to re-do?
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The analysis data is stored in the library, and by default in the new tags (and some compatibility tags which are auto-generated). These are stored in file tags, so an Update Library (from tags) would re-read them:
MC Tags:
Dynamic Range (DR)
Dynamic Range (R128)
Peak Level (R128)
Peak Level (Sample)
Volume Level (R128)
Volume Level (ReplayGain)
Compatibility Tags:
REPLAYGAIN_TRACK_GAIN
REPLAYGAIN_TRACK_PEAK
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Thanks, MrC!
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MC19 makes several important improvements to audio analysis and volume leveling:
- Adoption of the R128 industry standard (http://tech.ebu.ch/loudness) to analyze the loudness and dynamic range of content
- Ability to analyze audio for video files, including surround sound
- Smarter Volume Leveling that automatically respects intentional between track levels when playing from an album
- Volume Leveling works together with Adaptive Volume's peak level normalization (http://yabb.jriver.com/interact/index.php?topic=82023.0)
- Peak level is reported in decibels, measured as an R128 compliant True Peak, and reported per channel
This is a big topic, so I'll try to revisit it and post more details about the changes and the motivation at a later date.
Hi,
This is another very good and usefull addition in MC, and still under development I hope. During the development please make sure Stereo Playback from Video is taken care off. To me it seems that the impact on Stereo Video playback is huge. If I play 24/96 Audio files from CD I have an Audiophile performance, when playing this kind of format from Music Video it's not (dynamics are gone) Playing around with Adaptive Volume does not do much good. So my request, please make sure that Stereo Streams from Video are handled without tweaking, I would like it pure, Audiophile....
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This is another very good and usefull addition in MC, and still under development I hope. During the development please make sure Stereo Playback from Video is taken care off. To me it seems that the impact on Stereo Video playback is huge. If I play 24/96 Audio files from CD I have an Audiophile performance, when playing this kind of format from Music Video it's not (dynamics are gone) Playing around with Adaptive Volume does not do much good. So my request, please make sure that Stereo Streams from Video are handled without tweaking, I would like it pure, Audiophile....
Check the Dynamic Range (DR) and Dynamic Range (R128) values for both files.
What you are probably hearing, is exactly what R128 was designed to do. R128 plays back both files at the same volume level, regardless of their dynamic range.
Previously, tracks with a compressed dynamic range (a lot of modern music) would sound a lot louder than well mastered tracks with a lot of dynamic range.
Subjectively, most people will pick the louder of two tracks as sounding "better" regardless of how it was mastered.
When you eliminate the difference in volume between the two, you can now identify which track is more dynamic.
So the move towards R128 in broadcast now discourages the use of compressors when mastering music, because all it does is make your music sound worse, rather than making a trade-off between dynamics and loudness. (because you don't get the loudness benefit from it any more)
And if you just want to play videos at full volume with volume leveling disabled, you can set up a separate zone for them using Zone Switch, and disable volume leveling for it.
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Hi,
Is this feature can be used (understand: compatible with) with a room correction system placed after my MC computer ?
When I says "room correction system", I'm thinking about Trinnov or Audyssey.
Is there any link between adaptative volume and the audio analysis feature (R128) ? Is the same "thing" ?
I can not tell the difference between both terms.
Regards.
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Is this feature can be used (understand: compatible with) with a room correction system placed after my MC computer ?
When I says "room correction system", I'm thinking about Trinnov or Audyssey.
I don't see any reason it would not work.
Is there any link between adaptative volume and the audio analysis feature (R128) ? Is the same "thing" ?
I can not tell the difference between both terms.
Volume Leveling will adjust the volume of analyzed files during playback so that the average volume is the same. (with a fixed target of -23 LUFS average volume)
If you have a mixed playlist with tracks from lots of different albums, it will mean that the volume should not change between the tracks as it would without volume leveling.
Files from the same album will have one adjustment for all tracks. (to keep intended differences in volume between those tracks)
Adaptive Volume tries to make playback as loud as possible. If volume leveling is not enabled, tracks will be adjusted individually to be as loud as possible, so there may be large variances in volume between each track.
If your files are analyzed, adaptive volume can be combined with volume leveling. This means that all tracks in the current playlist are equal in volume, but the playlist itself is made as loud as possible. (while still keeping the tracks at the same volume)
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I don't see any reason it would not work.
Volume Leveling will adjust the volume of analyzed files during playback so that the average volume is the same. (with a fixed target of -23 LUFS average volume)
If you have a mixed playlist with tracks from lots of different albums, it will mean that the volume should not change between the tracks as it would without volume leveling.
Files from the same album will have one adjustment for all tracks. (to keep intended differences in volume between those tracks)
Adaptive Volume tries to make playback as loud as possible. If volume leveling is not enabled, tracks will be adjusted individually to be as loud as possible, so there may be large variances in volume between each track.
If your files are analyzed, adaptive volume can be combined with volume leveling. This means that all tracks in the current playlist are equal in volume, but the playlist itself is made as loud as possible. (while still keeping the tracks at the same volume)
Thanks for letting me know. Will try to play with this.
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So my request, please make sure that Stereo Streams from Video are handled without tweaking, I would like it pure, Audiophile....
All these features are optional. If you want it "pure", then just turn it off, and you get the audio as it was encoded in the file.
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If volume leveling only or volume leveling + adaptative volume are enabled, is MC internal volume disabled ?
So that means the only way to play louder or quiet is to deal with DAC volume.
I have some difficulties to understand in wich case I have to use this way or this way and what is the better way...
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Just finished the re-analysis of my entire library. Went faster than I expected.
I do have about 15 files that will not analyze. They just stay at 0% forever and while MC still plays, it's basically locked up and I have to kill it.
Does anyone at JRiver want a copy of one of these audio files to test with?
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What you are probably hearing, is exactly what R128 was designed to do. R128 plays back both files at the same volume level, regardless of their dynamic range.
Previously, tracks with a compressed dynamic range (a lot of modern music) would sound a lot louder than well mastered tracks with a lot of dynamic range.
Subjectively, most people will pick the louder of two tracks as sounding "better" regardless of how it was mastered.
When you eliminate the difference in volume between the two, you can now identify which track is more dynamic.
So the move towards R128 in broadcast now discourages the use of compressors when mastering music, because all it does is make your music sound worse, rather than making a trade-off between dynamics and loudness. (because you don't get the loudness benefit from it any more)
This is very well put.
The same goes for moving to R128 in software like JRiver (and hopefully all others eventually). It's a way to fight the loudness war.
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Hi,
I tried both volume adaptive + volume leveling after having analyzed all my audio files (9k in 2 hours).
But I'm not able to hear a difference with and without. And some audio tracks play louder than others...So I suspect it doesn't work at all for me.
Any explanation ? How to test or verify ?
Thanks
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Just finished the re-analysis of my entire library. Went faster than I expected.
I do have about 15 files that will not analyze. They just stay at 0% forever and while MC still plays, it's basically locked up and I have to kill it.
Does anyone at JRiver want a copy of one of these audio files to test with?
Can you download AudioTester (http://vuplayer.com/other.php) and test the files? No install, single executable. Just start it and drag/drop the files onto it.
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Hi,
I tried both volume adaptive + volume leveling after having analyzed all my audio files (9k in 2 hours).
But I'm not able to hear a difference with and without. And some audio tracks play louder than others...So I suspect it doesn't work at all for me.
Any explanation ? How to test or verify ?
Thanks
Few checks you can do.
Check the tags for R128 to see if they have values.
Add a few files to playing now, start playing them and check audio path; hover the mouse over DSP button and see the popup, it shows input, changes and output. Under changes it should tell you volume leveling is used and by how much.
Open DSP Studio and while still playing the songs and disable/enable the volume leveling/adaptive volume options. Unless the changes from volume leveing are 0dB, you should here a difference I would think.
Obviously, DSP studio doesn't work when you're bitstreaming. In this case audio path is empty I think.
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When I ripped my CDs to the wav and flac using dBpoweramp, I checked the boxes to write track gain and write album gain on the DSP ReplayGain dialog box, but I did not check the box about R128. (I attached a screen cap that I hope shows up.) Anywho, given that JRiver v19 is now using the R128 standard to analyze loudness and dynamic range of content, does that mean that I should re-rip my CDs with the “EBU R 128 Calculated Gain” box checked in dbpoweramp? And if, god forbid, the answer to that question is yes, then should the “LUFS Target Volume” setting be left at -23 or changed to some other value?
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When I ripped my CDs to the wav and flac using dBpoweramp, I checked the boxes to write track gain and write album gain on the DSP ReplayGain dialog box, but I did not check the box about R128. (I attached a screen cap that I hope shows up.) Anywho, given that JRiver v19 is now using the R128 standard to analyze loudness and dynamic range of content, does that mean that I should re-rip my CDs with the “EBU R 128 Calculated Gain” box checked in dbpoweramp? And if, god forbid, the answer to that question is yes, then should the “LUFS Target Volume” setting be left at -23 or changed to some other value?
You just need to analyze the files with Media Center. Select any files that you want analyzed, right-click them and select Library Tools > Analyze Audio. Disable the "skip analyzed files" option if necessary. (I'm not sure if files created with dBpoweramp will require this option to be disabled)
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I have 80k+ songs. Will I need to re analyze all of these every time I add new music?
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No.
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I just installed mc19 a few days ago and re-analyzed all of my audio files, about 27k. I now have 2 sets of peak level, volume level & dynamic range results when i look at individual file tags. Is this normal. If not what should I do?!
Thx!
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Im just installed mc19 a few days ago and re-analized all of my audio files, about 27k. I now have 2 sets of peak level, volume level & dynamic range results when i look at individual file tags. Is this normal. If not what should I do?!
This is normal. Media Center uses the Volume Level (R128) tags during playback.
The Volume Level (ReplayGain) tags are for compatibility with other players - Media Center can read legacy ReplayGain tags (which are not as good as R128) and after analysis will write ReplayGain 2.0 tags with the R128 data.
Dynamic Range (R128) is a measure of the track's overall dynamic range. (difference between loudest and quietest parts)
Dynamic Range (DR) is a measure of crest factor (https://en.wikipedia.org/wiki/Crest_factor) and produces results compatible with the Dynamic Range Database (http://www.dr.loudness-war.info/).
Tracks with higher dynamic range values are typically better mastered than those with lower dynamic range values.
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Makes sense now. Thx!!
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No.
But I do need to analyze the new tracks, right?
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But I do need to analyze the new tracks, right?
You will. Auto-import can be set to automatically analyze new files though - but it may skip over files if they have existing ReplayGain data. (I'm not sure if that applies to MC19 now due to the move to R128)
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Would someone please be so kind as to clarify the meaning of [and distinction between]:
"Write tags when analyzing audio..." (options / library & folders / configure auto-import)
&
"Update tags when file info changes" (options / general / importing & tagging)
Is there any dependency between these two settings?
Clarification would be appreciated - thanks.
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So it took three days to analyze my files, but it's done. The result? It works, sure, but the overall volume is WAY lower. I mean, I have to turn my system up significantly to get the same volume I have without the leveling. Is this normal?
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So it took three days to analyze my files, but it's done. The result? It works, sure, but the overall volume is WAY lower. I mean, I have to turn my system up significantly to get the same volume I have without the leveling. Is this normal?
Yes, audio is normalized to a target of -23 LUFS. (dB)
If you did not reduce the volume this much, you would have a problem with the volume level changing, as very dynamic tracks would hit the clipping protection.
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Does MC19 still support ReplayGain tags?
I do not believe we got a clear answer on this question? I am getting ready to install a build of v19 soon - but risk any damage to my library (the rest of household will be on v18 for some time yet) if v19 does something to existing RG tags.
So..
1. Does v19 do anything to existing RG tags in files analyzed via v18?
2. Can I install v19 and continue to use all RG data as if the R128 goodies do not exist?
VP
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Have been analyzing for 3 days and have another 3 or 4 days to go. Here is an interim report.
Most dvds seem to analyze ok but I had a couple for which the status ran up to something like 200,000% before it completed successfully and changed to Done.
Am collecting a variety of audio and video files that error out when analyzed. Expect 50 or so out of 180,000 so not a problem that I care gets fixed. Does JRiver wish to chase these oddballs down or shall I drop the issue?
I did come across one mp3 album that consistently causes MC to crash during analysis. We should definitely chase this down but I will wait to see if there are any more like this before providing details.
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I have two versions of Dire Straits / Brothers In Arms.
One version is the 2013 MFSL remaster and the other 24 bit 96khz.
Using "One World" as an example.
Dynamic Range
MFSL: 17
24bit96kHz: 11
The 24bit96kHz definitely sounds "louder" with volume leveling active (adaptive volume: OFF)
I don't have MC18 installed any longer so I can't compare to the old ReplyGain volume leveling.
It's not OMG louder, but it's noticeable when switching back and forth between tracks.
Is this normal for "Volume Leveling" when the DR is different?
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Have been analyzing for 3 days and have another 3 or 4 days to go. Here is an interim report.
Most dvds seem to analyze ok but I had a couple for which the status ran up to something like 200,000% before it completed successfully and changed to Done.
Am collecting a variety of audio and video files that error out when analyzed. Expect 50 or so out of 180,000 so not a problem that I care gets fixed. Does JRiver wish to chase these oddballs down or shall I drop the issue?
I did come across one mp3 album that consistently causes MC to crash during analysis. We should definitely chase this down but I will wait to see if there are any more like this before providing details.
It would probably be good if you could send samples to the JRiver team.
I have two versions of Dire Straits / Brothers In Arms.
One version is the 2013 MFSL remaster and the other 24 bit 96khz.
Using "One World" as an example.
Dynamic Range
MFSL: 17
24bit96kHz: 11
The 24bit96kHz definitely sounds "louder" with volume leveling active (adaptive volume: OFF)
I don't have MC18 installed any longer so I can't compare to the old ReplyGain volume leveling.
It's not OMG louder, but it's noticeable when switching back and forth between tracks.
Is this normal for "Volume Leveling" when the DR is different?
There are a couple of possibilities here.
1. Depending on how the albums are tagged, Media Center may be inadvertently treating both tracks as being from the same album, so they have a fixed correction. To check this, only have both versions of the same track in the "now playing" view, with the Volume Level (R128) information visible. Check that the tracks are using the correct level in the Audio Path window.
In my library, I use the Description field to specify the version of an album, which means that Media Center thinks they're from the same album, because its name is the same:
(http://www.abload.de/img/volume-levelgfqos.gif)
As you can see, when I add the CD track, the volume level for both is changed to −10 dB, even though the currently playing track should be +0.9 dB.
This results in the much better quality SACD track being played back 10.9 dB quieter than the CD one.
2. It's possible that the difference in dynamic range is large enough, that it's not possible for both tracks to perceptually sound the same. R128 does a really good job—better than ReplayGain ever did—but it may not be possible to level things exactly.
3. It may also be that you have run out of headroom, and require more than −23 LUFS allows for. (this is something that will hopefully be addressed in an update)
In your "now playing" view, add an expression column using:
Delimit(if(isempty([Peak Level (R128)]),,formatnumber(math(removecharacters(left([Peak Level (R128)],5),/ /+,0)+RemoveCharacters([Volume Level (R128)],/ LU,0)+1),1)),/ dB,)If this is a positive number for any of the tracks, it means that more headroom is required for proper volume leveling.
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1. Depending on how the albums are tagged, Media Center may be inadvertently treating both tracks as being from the same album, so they have a fixed correction. To check this, only have both versions of the same track in the "now playing" view, with the Volume Level (R128) information visible. Check that the tracks are using the correct level in the Audio Path window.
Both albums had slightly different album names.
2. It's possible that the difference in dynamic range is large enough, that it's not possible for both tracks to perceptually sound the same. R128 does a really good job—better than ReplayGain ever did—but it may not be possible to level things exactly.
3. It may also be that you have run out of headroom, and require more than −23 LUFS allows for. (this is something that will hopefully be addressed in an update)
In your "now playing" view, add an expression column using:
Delimit(if(isempty([Peak Level (R128)]),,formatnumber(math(removecharacters(left([Peak Level (R128)],5),/ /+,0)+RemoveCharacters([Volume Level (R128)],/ LU,0)+1),1)),/ dB,)If this is a positive number for any of the tracks, it means that more headroom is required for proper volume leveling.
None are positive, but one was -2.7 ... (thanks for the expression)
I suspect as you say, it's just the limited dynamic range of the 24bit96kHz version compared to the MFSL version.
It really sucks that the people who do the "remastering" will limit the dynamic range. One simply can't know that a "24bit96kHz" album is any better than other versions. But LOL... I own the MFSL version and will be returning the 24bit96kHz version back to a friend... it was fun to compare. If there is a fidelity difference between these versions (other than DR) I can't hear it.
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Do you have very many real world examples in your library where the Peak Level (R128) doesn't have enough headroom to adjust for Volume Level (R128)? In my library of 6400 audio tracks I found just three and it only made a 1.6 dB or less difference. I also would never play back those tracks without the entire album so it wouldn't matter anyway. The one needing the most Volume Level (R128) measures 15.8 and it has a Peak Level (R128) of -15.1 dBTP. The actual adjustment is 14.1 dB.
Only 200 tracks of mine need a positive volume adjustment (.03%). Out of those 200, there are only 4 tracks that I would play in a mixed playlist.
None of mine are >= 0, but I have some that are -0.1
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3. It may also be that you have run out of headroom, and require more than −23 LUFS allows for. (this is something that will hopefully be addressed in an update)
Is this the same issue that may cause some music to clip?
I have one song (at least one that I noticed last night) that was hitting 100% when viewing the DSP window in setup. The only thing I had checked was Up-sampling, so I hit STOP, unchecked up-sampling, started the song again, and it still hit 100% quite often.
Running 19.0.38 with volume set to Internal.
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Is this the same issue that may cause some music to clip?
No, Volume Leveling should not allow music to clip - it simply means that the track will be played quieter than "ideal" for leveling due to clipping protection.
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No, Volume Leveling should not allow music to clip - it simply means that the track will be played quieter than "ideal" for leveling due to clipping protection.
I thought I also tried it with Volume Leveling, but will double-check that tonight to be sure. I report back with my findings....now I need to remember the song - LOL
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Mr.C. "The analysis data is stored in the library, and by default in the new tags (and some compatibility tags which are auto-generated). These are stored in file tags, so an Update Library (from tags) would re-read them:"
I got a question, I'm backing up my files extern about 2.5 Gb. I did analyze my music files , some backup programs with incremental sync.
don't write the DR 128 in my backup files, like Microsoft Richcopy. Does the quote means that in case of HDD failure and restore my library I can Update my library from restore library all my DR 128 taggs are there again or not? Or should make a complete new backup.
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I got a question, I'm backing up my files extern about 2.5 Gb. I did analyze my music files , some backup programs with incremental sync.
don't write the DR 128 in my backup files, like Microsoft Richcopy. Does the quote means that in case of HDD failure and restore my library I can Update my library from restore library all my DR 128 taggs are there again or not? Or should make a complete new backup.
The library or a library backup contains all of your data (except the actual media files and cover art). The file's tags are essentially just a backup and also useful if you move files to another program.
In other words, if you have the library, you have the audio analysis data and can always apply it to the tags at a later date.
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Thanks Matt, that's wonderful and a great future.
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Also new to this and have 2 questions.
1. Will analyzing my audio change the files in any way or is the data stored outside of the file. Just wondering because I have my music sync'd to a drive on a server to keep it safe. Don't mind if its going to have to resync the whole thing again.
2. How will this effect mixed albums that come with a .cue sheet. Example I have an album that's 2 hours long but in a single .FLAC but with a .cue file to break up each track in JRiver. So how would that work to normalize the album?
EDIT. Decided to take the plunge and let it plow through the 28,000 audio files.
I am curious though does it have to read the entire file to do what its doing or how exactly does that work. Same for video is it going to have to read a 4-16gb file just to do its job?
Also notice it added and or modified the BPM field of the id3 tags so never mind the first question.
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1. Will analyzing my audio change the files in any way or is the data stored outside of the file. Just wondering because I have my music sync'd to a drive on a server to keep it safe. Don't mind if its going to have to resync the whole thing again.
By default the audio analysis results are stored in the library AND file tags. If this bothers you, you can turn off writing of any field to the tag using Options > Library & Folders > Manage Library Fields. I would not recommend doing this.
I am curious though does it have to read the entire file to do what its doing or how exactly does that work. Same for video is it going to have to read a 4-16gb file just to do its job?
It has to read and decode the entire file to analyze all the audio. It's like playing, only at full speed instead of 1x.
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Just wanted to say thank you for these features. I notice that I no longer need to adjust the volume, even on playlists that include files like old 64kb mono tunes from early jazz 78's. Amazing! I re-analyzed my 20K songs the other day, and was listening with the old Sennheiser HD-580's last night, and I'm convinced that even the mp3's sound more transparent, crisp and defined than they used to. :P
Will this hold true when playing to a tab via gizmo? My Asus died yesterday, so I'm shopping for another (don't really mind ;) ), and I stream my workout mix back to the 'gym' - have noticed major vol. fluctuations in the past. Thanks, again.
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I'm convinced that even the mp3's sound more transparent, crisp and defined than they used to.
They may very well do, as Volume Leveling should help avoid inter-sample clipping, which is common with lossy formats.
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I have been noticing quite a reduced volume in .dff files and I have 4 copies of that Jones album so I wonder if it is an issue? Doesn't DSD have -6db or something? But these ISO are being converted to DXD on fly by MC. I'll have to look at the tags tonight.
Anyway, can someone comment on best settings for using RS128 with a DAC with digital volume control? If I want to do most volume adjustment in PC, and minimize use in the DAC (Auralic Vega). What is Volume Disabled vs. Internal Volume with regard to using RS128?
thanks!
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Guys,
Thanks for rthe feature.
Does enabling just levelling hurt dynamics and sq?
If yes, is there anything that can be done about it?
If I disable I internal volume, so its always @100%, does this feature still work? I ask because I'm getting an analogue preamp installed and would like to us purely that.
Thanks for your help.
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Does enabling just levelling hurt dynamics and sq?
As long as your audio device is at least 24-bit, it should not hurt sound quality.
If anything, using volume leveling should improve your perception of dynamics.
If I disable I internal volume, so its always @100%, does this feature still work? I ask because I'm getting an analogue preamp installed and would like to us purely that.
Yes, Volume Leveling will still work.
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I previously asked on a different thread if volume levelling will affect quality. I should have done the test , being lazy !
I did the test , I used 2 of my "test tracks" , Joni Mitchell - Blue - A Case of You , and Dire Straits - Love Over Gold - Private Investigation both FLAC via MC from remastered CD's
Out via ASIO to a Cambridge Audio DAC Magic , then Denon AV amp (set Direct) and HD800 .
Even to my aging ears , I believe I can tell a difference on the tops , sibilance as less pronounced and the aggressive "twang" of the guitar in Dire Straits was definitely missing.
I saw a dramatic reduction in volume of some tracks , presumably because I have a very high level track or two in the library
Are there settings to adjust ?. I have switched it off for now
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I previously asked on a different thread if volume levelling will affect quality. I should have done the test , being lazy !
I did the test , I used 2 of my "test tracks" , Joni Mitchell - Blue - A Case of You , and Dire Straits - Love Over Gold - Private Investigation both FLAC via MC from remastered CD's
Out via ASIO to a Cambridge Audio DAC Magic , then Denon AV amp (set Direct) and HD800 .
Even to my aging ears , I believe I can tell a difference on the tops , sibilance as less pronounced and the aggressive "twang" of the guitar in Dire Straits was definitely missing.
I saw a dramatic reduction in volume of some tracks , presumably because I have a very high level track or two in the library
Are there settings to adjust ?. I have switched it off for now
Volume leveling works by (mostly) reducing the volume of tracks to recenter them to a specific standard level. The result is that (with leveling on) most music will sound about the same volume regardless of compression (i.e. heavily compressed music will not "sound louder" due to compression). Generally speaking, louder music (even slightly louder music) sounds "better" to most people, which is how the loudness wars got started to begin with.
You don't mention whether you readjusted the volume in your experiment so that you were listening at the same "effective" volume level in both cases, but, if you haven't, I would recommend doing that before coming to any firm conclusions about volume leveling.
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Even to my aging ears , I believe I can tell a difference on the tops , sibilance as less pronounced and the aggressive "twang" of the guitar in Dire Straits was definitely missing.
If you add an expression column with the following code, are there any values above 0?
math(removecharacters(listitem([Peak Level (R128)],0),+/ dBTP,0)+1)
If they are above 0, than means what you're probably hearing is the removal of inter-sample clipping, which is a very bad artifact of modern mastering processes.
If not, you're just hearing a difference in volume.
I saw a dramatic reduction in volume of some tracks , presumably because I have a very high level track or two in the library
Everything is normalized to -23 dB, so it depends how loud the track was mastered.
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Hi
While not "metering" the comparison to ensure similar volume levels in my head phones , I did the adjustment so to my ears they both sounded the same volume , there was not a massive difference
I created the Expression with the maths you provided and is see only zero values for the > 30,000 tracks in my library , is that to be expected ?
Mike
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I created the Expression with the maths you provided and is see only zero values for the > 30,000 tracks in my library , is that to be expected ?
Have you analyzed your library in MC19 yet?
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Hi,
What is the behavior when there are several audio tracks (languages) on a video ?
Which one is the reference ? Or maybe they are all analized ?
Regards.
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Hi,
What is the behavior when there are several audio tracks (languages) on a video ?
Which one is the reference ? Or maybe they are all analized ?
Regards.
The one that is analyzed is the one that is played when you select the movie. You can change the audio track during playback, then analyze after playback has stopped. The last used audio track will be analyzed.
If you want/need several tracks analyzed, the create a Particle of the movie and select a different track for the Particle.
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What is the behavior when there are several audio tracks (languages) on a video ?
The analyzer will analyze the current audio track (the one you would hear if you played the file).
You can configure how default audio tracks are selected in Options > Video.
Or you can play a video and change the audio track so that the current audio track is changed (for analysis, serving to DLNA, Gizmo, etc.).
(I cross-posted with mojave. As usual, he's faster than me.)
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One more guy who does it just like Jack, BigCat etc. I always use MP3Gain first and then MC. Would be great if I could throw MP3Gain away and use just MC for this purpose. Remember, like Jack said: Will MC19 be able to modify the files so they can play normalized outside of MC?
I asked this question a couple of months ago and for the answer I was referred to another topic. However, I'm sorry to say: I still can't figure out what the answer to my question is. Must be because English is not my native language - sorry for that.
So let me rephrase my question, it's actually a simple one:
If I use MC's new analyze-tool for all my mp3's, will these files have the same volume outside of MC (for instance in my car)? If yes, that would be great - then I can throw good old MP3Gain away.
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If I use MC's new analyze-tool for all my mp3's, will these files have the same volume outside of MC (for instance in my car)? If yes, that would be great - then I can throw good old MP3Gain away.
No. Since MC does not alter the physical makeup of the MP3 file.
Whether using the older ReplayGain with v18 or prior - OR the new system within v19 - the end result is the same - MC writes it's volume/level data to tags within the files themselves. That tag data can then be picked up by apps (or devices) that can handle ReplayGain or similar - but if you car has no ability to interpret these tags - then the file will playback as originally rendered.
Hope that helps.
VP
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No. Since MC does not alter the physical makeup of the MP3 file.
Whether using the older ReplayGain with v18 or prior - OR the new system within v19 - the end result is the same - MC writes it's volume/level data to tags within the files themselves. That tag data can then be picked up by apps (or devices) that can handle ReplayGain or similar - but if you car has no ability to interpret these tags - then the file will playback as originally rendered.
Hope that helps.
VP
Thanks VP. I guess I am stuck to MP3Gain, because in my car I play the mp3's from USB-stick and the player itself is not that fancy. Bummer..
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Thanks VP. I guess I am stuck to MP3Gain, because in my car I play the mp3's from USB-stick and the player itself is not that fancy. Bummer..
When writing your files to a handheld device (MP3 player or USB stick), you can tell JRiver to "apply DSP" to the file (including volume leveling). So if your goal is to modify the files that you write to a portable device to all be roughly the same volume, you can do that with JRiver (I just did it myself over the weekend for a long car trip). It's just not part of the normal file analysis process (to avoid altering the files).
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When writing your files to a handheld device (MP3 player or USB stick), you can tell JRiver to "apply DSP" to the file (including volume leveling). So if your goal is to modify the files that you write to a portable device to all be roughly the same volume, you can do that with JRiver (I just did it myself over the weekend for a long car trip). It's just not part of the normal file analysis process (to avoid altering the files).
That's right. Completely forgot about that!
VP
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When writing your files to a handheld device (MP3 player or USB stick), you can tell JRiver to "apply DSP" to the file (including volume leveling). So if your goal is to modify the files that you write to a portable device to all be roughly the same volume, you can do that with JRiver (I just did it myself over the weekend for a long car trip). It's just not part of the normal file analysis process (to avoid altering the files).
Now that sounds exactly what I want! However, where is this 'appy DSP' option situated? I always just use the 'Rename, move & copy files'-option to copy files to my USB-stick. But there is no DSP-thing there.
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Now that sounds exactly what I want! However, where is this 'appy DSP' option situated? I always just use the 'Rename, move & copy files'-option to copy files to my USB-stick. But there is no DSP-thing there.
I am now confused as well. I always thought you could only use the "Apply DSP" when burning a CD or using the Convert Format process...
I do not understand how one can "write DSP" to a file simply by moving it to a portable device. I load my iPod all the time but have never seen an "apply DSP" option.
Maybe mwillems can enlighten us?
VP
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I am now confused as well. I always thought you could only use the "Apply DSP" when burning a CD or using the Convert Format process...
I do not understand how one can "write DSP" to a file simply by moving it to a portable device. I load my iPod all the time but have never seen an "apply DSP" option.
Maybe mwillems can enlighten us?
VP
The trick is to tell JRiver that the usb stick is a handheld device. I'm not near JRiver right now so I can't describe the exact menu sequence, but in the options for configuring handheld devices there's an option to specify a file path as a handheld device. The option probably exists because some MP3 players aren't detected as MP3 players and just show up as a drive (e.g. "N:\"). So you have to go into that option list and specify the drive letter for your USB stick as a handheld. At that point JRiver will treat it as an MP3 player that can be synced. You can then set up which files you want to sync through a number of different methods (drag and drop, send to, creating a playlist, etc.). Once you've lined up all the files you want to transfer, you need to go into "sync options" or "sync details" before you press the sync button. In that options menu there is an "apply DSP" checkbox that will apply DSP (including volume leveling) to the files during the file transfer to the device.
It may sound complicated, but if you've used JRiver with a handheld device before it should be pretty easy. I originally started using it before "apply DSP" was a thing to manually back up files to an external harddrive. By configuring the harddrive as a handheld, I can take advantage of JRiver's "only sync files that have been changed" function to reduce the amount of file transfer that needs to happen for the manual backup.
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Hi Matt,
I am using current version of Media Center 19.0.66.
I think volume leveling is not working as expected. Now I have the album "Muddy Waters" - "Folk Singers" in two Versions. First one is an EAC CD Rip in 16Bit/44Khz FLAC. The other, newer one is 24Bit/192Khz AIFF from HD Tracks. Both albums analyzed with JRiver. Putting the same single Track from each album to a Playlist named "TestGain"). Adaptive Volume (Peak Level Normalize) and Volume Leveling is enabled.
- On playback the track form CD sounds much louder then track from HD Tracks. Why?
- And why is volume leveling in Audio Path the same for both tracks?
Tags for Track from CD:
Free Lossless Audio Codec (FLAC)
44,1 kHz Sample Rate
16 Bits Per Sample
2 Channels
Min Block Size = 4096
Max Block Size = 4096
Min Frame Size = 725
Max Frame Size = 11669
Padding Block = 7820 bytes
# of Pictures = 1
Type 3 (21529 bytes)
SeekTable Block (576 bytes):
# of points = 32
Vorbis Comment Block (621 bytes):
Vendor String = S
TOTALDISCS = 1
TOTALTRACKS = 14
TOOL NAME = Media Center
TOOL VERSION = 19.0.66
REPLAYGAIN_TRACK_GAIN = +0.52 dB
BPM = 62
VOLUME LEVEL (R128) = -4,480410099029541
PEAK LEVEL (R128) = -0,1 dBTP; -0,8 Left; -0,1 Right
REPLAYGAIN_TRACK_PEAK = 0.977000
TRACKNUMBER = 3
DATE = 1964
GENRE = Blues
DYNAMIC RANGE (R128) = 18,7092399597167969
ALBUM = Folk Singer
TITLE = My Captain
RATING = 4
PEAK LEVEL (SAMPLE) = -0,2 dB; -0,8 Left; -0,2 Right
DYNAMIC RANGE (DR) = 14
VOLUME LEVEL (REPLAYGAIN) = 0,5195900201797485
DISCNUMBER = 1
COMMENT = EAC
ARTIST = Muddy Waters
Audiopath on playback:
(http://abload.de/img/audiopathtrackfromcda5ury.jpg) (http://abload.de/image.php?img=audiopathtrackfromcda5ury.jpg)
Tags for Track from HDTracks:
Uncompressed Audio File (aif)
192,0 kHz, 24 bit, 2 ch
ID3v2.3 Tag: (524288 bytes)
TIT2 (Name): My Captain
TPE1 (Artist): Muddy Waters
TPE2 (Album Artist): Muddy Waters
TALB (Album): Folk Singer
TRCK (Track #): 3
TYER (Year): 1964
TCON (Genre): Blues
TCOM (Composer): Willie Dixon
TBPM (BPM): 63
APIC (Picture) (Cover): <too large to display>
TXXX (Album Artist): Muddy Waters
TXXX (Dynamic Range (DR)): 17
TXXX (Dynamic Range (R128)): 19,3680591583251953
TXXX (Peak Level (R128)): -3,4 dBTP; -4,0 Left; -3,4 Right
TXXX (Peak Level (Sample)): -3,4 dB; -4,0 Left; -3,4 Right
TXXX (replaygain_track_g..): +7.61 dB
TXXX (replaygain_track_p..): 0.676000
TXXX (Tool Name): Media Center
TXXX (Tool Version): 19.0.66
TXXX (Volume Level (R128)): 2,6068499088287354
TXXX (Volume Level (Repl..): 7,6068501472473145
Audiopath on playback:
(http://abload.de/img/audiopathtrackfromhdthcr2z.jpg) (http://abload.de/image.php?img=audiopathtrackfromhdthcr2z.jpg)
should i send you the two tracks for testing to logs at jriver dot com?
Thanks Erwin
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Adaptive Volume (Peak Level Normalize) and Volume Leveling is enabled.
I will let Matt tackle this one - but why do you have Peak Level Normalize on? That is pushing the level up as high as it can go without clipping (I believe)...
"If you use Volume Leveling AND Peak Level Normalization, you will get the loudest playback of the current playlist that maintains equal volume between tracks and prevents all clipping. The volume between tracks in the playlist will be the same, but the volume between different playlists could be different (since each playlist will have a different peak level normalization value)."
I only have Volume Levelling engaged in my current testing of v19 and that Muddy Waters records sounds fantastic.
VP
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but why do you have Peak Level Normalize on? That is pushing the level up as high as it can go without clipping (I believe)...
Wy not? The reason is what you have said: 2That is pushing the level up as high as it can go without clipping"
There should be no SQ problem with this option, right?
Now I tested playback without Peak Level Normalize. Same problem: CD Track sounds much louder, and Audiopath shows same value (- 4,5 dB) for volume leveling. shouldend it be different for playing the different files with the same loudness?
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Why not? The reason is what you have said: That is pushing the level up as high as it can go without clipping" There should be no SQ problem with this option, right?
Pushing the level up as high as it can go without clipping - is probably not what Muddy (or any other artist ever intended). SQ quality will be effected since playback can and will be unnatural with some material.
Adaptive Volume was really designed for playlists where you could have tracks that have wide volume swings. It then glues together a playlist volume wise BUT at the expense of cranking ALL tracks to the highest peak value (max). Kinda like listening to a poorly mastered 2013 pop album on your iPod where the volume is goosed to the max and every track is at the same relentless volume. Of course Adaptive Volume has it's place if you were in a car, a party or other noisy environment where you having everything "leveled" to the max would be beneficial. Read here for more..
http://yabb.jriver.com/interact/index.php?topic=82023.0
While everyone has their own tastes - for me personally - I would never "normalize" anything. Just let the natural dynamics of the source material do the talking - and the new Volume Leveling and R128 enhancements do an excellent job of that in v19 on their own without resorting to Peak Level Normalize..
Cheers,
VP
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Pushing the level up as high as it can go without clipping - is probably not what Muddy (or any other artist ever intended). SQ quality will be effected since playback can and will be unnatural with some material.
I think you might be thinking of dynamic range compression. That does not apply here.
Faster's screenshots say "Peak level normalize (fixed)". This means it's no different than turning the volume up (or down) at the beginning of the song. I think Muddy would be happy to let you listen at any volume :)
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should i send you the two tracks for testing to logs at jriver dot com?
Yes, please.
I can't explain why the second track (aif) is showing -4.5dB for volume leveling.
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I think you might be thinking of dynamic range compression. That does not apply here.
I guess I am misunderstanding how Adaptive Volume works then. Will study some more.
Cheers!
VP
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Since both songs have the same album tag there won't be an adjustment since "intentional volume differences between tracks on the same album are preserved." It doesn't matter that both songs are from different labels.
Regarding Peak Level Normalize - it adds back in the lost volume due to volume leveling. It rarely increases it even further. For example, if an album is played that has -4.5 dB in the Volume Level (R128) tag, the volume will be reduced by 4.5 dB when just that album is added to the playlist and Volume Leveling is on. If you also turn on Peak Level Normalize, then 4.5 dB will be added back in for a net change of 0 dB. Since the album is the only one in the playlist, its volume obviously does not need to be reduced for Volume Leveling. Basically Adaptive Volume (Peak Level Normalize) intelligently looks at the items in the playlist and uses the least amount of volume reduction necessary to still achieve Volume Leveling.
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Since both songs have the same album tag there won't be an adjustment since "intentional volume differences between tracks on the same album are preserved." It doesn't matter that both songs are from different labels.
This.
Many of us have different versions of the same album (Dark Side of the Moon, I think I have like 12 versions) and volume leveling doesn't consider [fields] other than album, artist.
@Faster: As a test, just name one of the albums "Folk Singer (CD)" and test again.
PS. Try and find the the MFSL UDCD-593 version ;).
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Thanks guys for pointing out that the two files were on the same album. That had flown right over my head, and I'm sure that's the explanation.
If they're different albums, I'd recommend giving them different names. We might also be able to leverage the album analyzer which considers file path, but this opens up a new set of problems with users that don't reliably use album folders.
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I use the Publisher tag to differentiate albums, but I don't know if that would be standard with other users.
I really can't imagine that anyone listens to the same song from different versions of the same album in one playlist. ? I think this was just for testing.
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I guess I am misunderstanding how Adaptive Volume works then. Will study some more.
Hopefully this will help:
http://yabb.jriver.com/interact/index.php?topic=82023.0
The peak level normalize mode of Adaptive Volume does no dynamic range compression. For audio, it's mostly adding back the volume you'd lose from Volume Leveling.
The other two modes of Adaptive Volume do dynamic range compression, but the user interface and link above try to explain why you might want that in some cases.
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The peak level normalize mode of Adaptive Volume does no dynamic range compression. For audio, it's mostly adding back the volume you'd lose from Volume Leveling.
Understood. I wasn't implying "compression" but it came out that way wit the iPod comparison. I understand "normalization" in the sense of determining a peak value and then cranking everything else up to that level.
But to me that's never a good thing (to me personally) - unless of course one has a special need for it - like a specific playlist, late night viewing etc. Even with this option available - I still use just Volume Levelling alone for all my playlists and they all sound great - especially with the R128 additions in v19.
VP
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I use the Publisher tag to differentiate albums, but I don't know if that would be standard with other users.
I really can't imagine that anyone listens to the same song from different versions of the same album in one playlist. ? I think this was just for testing.
Yeh, that's true. I use several custom fields, for instance Source (CD, HDTracks, SACD) and Album Version (MFSL, 1991 Remaster, or Cat Number).
It would be convenient if I didn't have to edit the album tag, which is also a give-away as to which version is playing when I'm switching back and forth. But as you say, its often only for testing so not that important.
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Since both songs have the same album tag there won't be an adjustment since "intentional volume differences between tracks on the same album are preserved." It doesn't matter that both songs are from different labels.
That's the case!
After i have renamed albumtag in one of the files everything works as expected!
Same loudness and adjust volume by -4,5 for CD Version and + 2,4 dB for the HD Tracks Version in audiopath.
Thanks!! Great and clever Community here, an of course a very great product!
Now i better understand how volume leveling and adaptive volume works!
And this (my) is a great example especially with different Masters (compressed vs better dynamic range recordings) from the same album.
Track "My Captain" on the CD Version is 14 (which is already well) vs HD Track is 17 (witch is exellent)!
btw. highest DR on this musthave album is track "Long Distance Call" with DR18 on the HD Tracks version (CD Version DR14).
Erwin
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I have the HD Tracks Elton John - Madman Across the Water. The Madman Across the Water track is DR8.
The MCA Records version I have is DR10.
Earlier this week I purchased the DJM Records 825 487-2 made in West Germany that is supposed to have it at DR13. I haven't received the disc yet.
It will be interesting to compare.
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Track "My Captain" on the CD Version is 14 (which is already well) vs HD Track is 17 (witch is exellent)!
btw. highest DR on this musthave album is track "Long Distance Call" with DR18 on the HD Tracks version (CD Version DR14).
Erwin
Glad to hear you figured it out.
Those DR numbers are the same as the MFSL version I mentioned. If the HD Tracks is a true high res version (ie, not upsampled) I might have to get it too.
I have the HD Tracks Elton John - Madman Across the Water. The Madman Across the Water track is DR8.
The MCA Records version I have is DR10.
Earlier this week I purchased the DJM Records 825 487-2 made in West Germany that is supposed to have it at DR13. I haven't received the disc yet.
It will be interesting to compare.
One of my favority EJ albums.
That number is better than the MFSL version I have. Not that means its better but its enough reason to look into it. I may have to get that too :D.
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I have the HD Tracks Elton John - Madman Across the Water. The Madman Across the Water track is DR8. The MCA Records version I have is DR10. Earlier this week I purchased the DJM Records 825 487-2 made in West Germany that is supposed to have it at DR13. I haven't received the disc yet. It will be interesting to compare.
I just analyzed my MFSL Madman (1989) CD...and I get DR 13 in 19.0.67
Anyhoo - that WG DJM is awesome. All of the mid 80's DJM pressings are a wonderful thing.
VP
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I think I would like to use this feature. (Improved audio analysis and volume leveling (R128))
However, I have two concerns:
1. That R128 volume leveling NOT alter dynamic range or alter sound quality in any other way, and
2. Do the calculated levels alter the actual file in any way, other than to create a tag value?
It sounds like the R128 method (volume leveling), without Peak Normalize is what I want to or do I have it backwards?
Sorry if this has already been answered, but the posts here seem somewhat contradictory.
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1. That R128 volume leveling NOT alter dynamic range or alter sound quality in any other way, and
2. Do the calculated levels alter the actual file in any way, other than to create a tag value?
Neither Volume leveling or Peak Level Normalize alter dynamic range.
No changes are made to the audio data in your files.
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It sounds like the R128 method (volume leveling), without Peak Normalize is what I want to or do I have it backwards?Sorry if this has already been answered, but the posts here seem somewhat contradictory.
This is exactly what you want.
And as Matt mentioned - "No audio is harmed during the making of this R128 volume analysis" :)
VP
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The Dynamic Range (R128) is the R128-derived Loudness Range (LRA).
Semantics discussion split here (http://yabb.jriver.com/interact/index.php?topic=84985.0).
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I have the HD Tracks Elton John - Madman Across the Water. The Madman Across the Water track is DR8.
The MCA Records version I have is DR10.
Earlier this week I purchased the DJM Records 825 487-2 made in West Germany that is supposed to have it at DR13. I haven't received the disc yet.
It will be interesting to compare.
Thanks, Matt, for incorporating ITU standard loudness normalization in JRiver. Another obstacle to ending the loudness race has fallen!
Let's stop using these non-standard terms that mean nothing. The so-called terms "DR" which have been coined by a non-standards organization mean nothing. They use it to describe the level, not the dynamic range. The proper internationally-standardized loudness measurement is called "LUFS", for loudness units full scale. As Matt described earlier in the thread, the proper term for "dynamic range" is NOT DR, but rather LRA, for loudness range. This is also a standardized measurement by the ITU.
I do not believe that LRA is a good measure of sound quality, however. There will be something better down the pike, but until then, don't try to rely on LRA measurements as a judgment of whether you should buy a recording or not. For example, a good Steely Dan might have a rather small LRA, but be a very open-sounding, uncompressed recording with clean transients. But Tool's Aenima is purposely highly compressed---it has a very poor transient response and the sound is squashed to death. But loudness range? it has plenty of it. How did they get that? By manipulating the gain in mastering AFTER the compressor was applied. So I can make a recording which has a ton of measurable range but still sounds squashed and compressed. That's why LRA is a poor measurement of sound quality.
Stick with LUFS and that's about as close as you can get to an idea of whether a pop recording is overcompressed. But still far from perfect. No time to get into that today!
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Thanks, Matt, for incorporating ITU standard loudness normalization in JRiver. Another obstacle to ending the loudness race has fallen!
Let's stop using these non-standard terms that mean nothing. The so-called terms "DR" which have been coined by a non-standards organization mean nothing. They use it to describe the level, not the dynamic range. The proper internationally-standardized loudness measurement is called "LUFS", for loudness units full scale. As Matt described earlier in the thread, the proper term for "dynamic range" is NOT DR, but rather LRA, for loudness range. This is also a standardized measurement by the ITU.
I made a big push for this on the beta testing forum. Matt felt that it would be easier for people to understand "Dynamic Range (R128)" than "Loudness Range (LRA)" or something similar to it.
We had the same argument about using LU rather than LUFS. I was also of the opinion that Media Center should match the ITU as much as possible.
I do not believe that LRA is a good measure of sound quality, however. There will be something better down the pike, but until then, don't try to rely on LRA measurements as a judgment of whether you should buy a recording or not. For example, a good Steely Dan might have a rather small LRA, but be a very open-sounding, uncompressed recording with clean transients. But Tool's Aenima is purposely highly compressed---it has a very poor transient response and the sound is squashed to death. But loudness range? it has plenty of it. How did they get that? By manipulating the gain in mastering AFTER the compressor was applied. So I can make a recording which has a ton of measurable range but still sounds squashed and compressed. That's why LRA is a poor measurement of sound quality.
I agree - the crest factor is generally a much better indicator of track quality than the loudness range - that's why we have both. Dynamic Range (DR) is a measure of crest factor, and the results should match the TT-DR meter.
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Let's stop using these non-standard terms that mean nothing. The so-called terms "DR" which have been coined by a non-standards organization mean nothing. They use it to describe the level, not the dynamic range.
From Sound On Sound's Sept. 2011 Article (http://www.soundonsound.com/sos/sep11/articles/loudness.htm) on Dynamic Range: "For instance, one criterion commonly used to describe the dynamic behaviour of a piece of recorded music is the 'crest' factor. Put simply, the crest factor is the difference between the RMS level and the peak level over the course of the song. Intuitively, it measures the amplitude of the emerging 'peaks' in the audio stream. It's considered a good marker of the amount of dynamic compression that was applied to the music: more compression generally means a lower crest factor. Some professionals consider good handling of the crest factor as the cornerstone of successful mastering. Also, still generally speaking, the lower the crest factor, the louder the music."
JRiver's Dynamic Range (DR) is a measure of the crest factor and, in my library, seems to represent compression or lack of compression quite well.
Here are the specifications (http://www.dynamicrange.de/sites/default/files/DR-Manual-V1_1-English.pdf) for Dynamic Range (DR) implemented by JRiver.
That's why LRA is a poor measurement of sound quality.
I haven't read any posts here of anyone that thinks LRA is a measure of sound quality.
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JRiver's Dynamic Range (DR) is a measure of the crest factor and, in my library, seems to represent compression or lack of compression quite well.
Despite the plentiful amount of measurement and analysis columns available in v19 - the TT-DR meter column (DR) is the only number I care about when considering a purchase or when considering inclusion into our library.
If that specific column value is painfully high (7,6,5,4) then regardless of what kind of sound quality the source has - it's totally irrelevant to me. Any source with a DR higher than maybe 9 (using TT-DR) gives me a headache after 2 tracks so it's immediately dismissed.
These DR numbers driven by TT have never failed me yet. I have yet to encountered ANY source file with a deafening DR (8,7,6,5) that either sounds good or is worth putting in the library and I have never heard a source file with a nice DR (14,13,12,11) that sounds bad or causes me to stop playback and reach for an Advil.
The R128 stuff is a technical dream to have in this class of program - but unless you know what all those columns mean in a professional sense - I think the TT-DR numbers are a nice easy way for Average Joe to instantly know if he's in the right ballpark and the source is not crushed to death.
VP
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Mojave. Using "DR" for crest factor is a stupid, misleading, misuse of the English language. For over 100 years, the term "dynamic range" has referred to the distance between loudest and softest passages, NOT the distance between average level (or loudness) and peak level. I thoroughly detest this reinvention and reuse of one term to substitute for another. I thoroughly endorse the use of a new term such as (invented by Thomas Lund) "PLR" for the quantity you wish to describe. PLR standing for "Peak to Loudness Ratio". But please cease and desist using the term "DR" to represent something which it is not. Thank you.
By the way, I'm glad that no one except some of those in the European Radio production community has mistakenly latched on LRA as a measure of sound quality. As you say, (but using a misleading term), PLR is a much better measure of sound quality. Let's call a spade a spade and tell the people at Sound On Sound and a few other places that they are opening a can of worms by trying to reapply "dynamic range" to another quantity.
BK
From Sound On Sound's Sept. 2011 Article (http://www.soundonsound.com/sos/sep11/articles/loudness.htm) on Dynamic Range: "For instance, one criterion commonly used to describe the dynamic behaviour of a piece of recorded music is the 'crest' factor. Put simply, the crest factor is the difference between the RMS level and the peak level over the course of the song. Intuitively, it measures the amplitude of the emerging 'peaks' in the audio stream. It's considered a good marker of the amount of dynamic compression that was applied to the music: more compression generally means a lower crest factor. Some professionals consider good handling of the crest factor as the cornerstone of successful mastering. Also, still generally speaking, the lower the crest factor, the louder the music."
JRiver's Dynamic Range (DR) is a measure of the crest factor and, in my library, seems to represent compression or lack of compression quite well.
Here are the specifications (http://www.dynamicrange.de/sites/default/files/DR-Manual-V1_1-English.pdf) for Dynamic Range (DR) implemented by JRiver.
I haven't read any posts here of anyone that thinks LRA is a measure of sound quality.
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Tell the people at TT to rename their misleading and incorrect terminology. Peak to loudness ratio is NOT dynamic range. And the TT meter is NOT a dynamic range meter.
BK
Despite the plentiful amount of measurement and analysis columns available in v19 - the TT-DR meter column (DR) is the only number I care about when considering a purchase or when considering inclusion into our library.
If that specific column value is painfully high (7,6,5,4) then regardless of what kind of sound quality the source has - it's totally irrelevant to me. Any source with a DR higher than maybe 9 (using TT-DR) gives me a headache after 2 tracks so it's immediately dismissed.
These DR numbers driven by TT have never failed me yet. I have yet to encountered ANY source file with a deafening DR (8,7,6,5) that either sounds good or is worth putting in the library and I have never heard a source file with a nice DR (14,13,12,11) that sounds bad or causes me to stop playback and reach for an Advil.
The R128 stuff is a technical dream to have in this class of program - but unless you know what all those columns mean in a professional sense - I think the TT-DR numbers are a nice easy way for Average Joe to instantly know if he's in the right ballpark and the source is not crushed to death.
VP
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My recommendation is that Matt truly stick to the ITU specs. ITU is an international standards organization. "DR" is not an internationally-recognized term. I recommend the term "PLR" or "Crest Factor" to apply to the distance between the average loudness and the highest measured peak of the material. It's not in the ITU book, but at least it won't confuse readers and users as DR clearly does.
The so-called "DR" meter is NOT a measure of dynamic range. Not in the least.
BK
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Tell the people at TT to rename their misleading and incorrect terminology. Peak to loudness ratio is NOT dynamic range. And the TT meter is NOT a dynamic range meter.
Bob,
100% agree with you.
However - since you are the "man" in some circles regarding this wide topic (I still read your "Mastering Audio" book with purpose :) - it would probably have more weight if you told the TT people to correct it. :)
Just sayin...
VP
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My recommendation is that Matt truly stick to the ITU specs. ITU is an international standards organization. "DR" is not an internationally-recognized term. I recommend the term "PLR" or "Crest Factor" to apply to the distance between the average loudness and the highest measured peak of the material. It's not in the ITU book, but at least it won't confuse readers and users as DR clearly does.
The so-called "DR" meter is NOT a measure of dynamic range. Not in the least.
I'd definitely like to see this change; I was never very keen on the "Dynamic Range (DR)" naming.
Is there a standard for this Peak to Loudness Ratio? I'm wondering if the Dynamic Range (DR) field should be renamed, or if it would be a new addition.
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I'd definitely like to see this change; I was never very keen on the "Dynamic Range (DR)" naming.
Is there a standard for this Peak to Loudness Ratio? I'm wondering if the Dynamic Range (DR) field should be renamed, or if it would be a new addition.
There is no official standard yet for the "PLR". But it is becoming a de facto standard being pushed by Thomas Lund of TC Electronic. It is the simple difference between the ITU highest true peak and the average loudness so it is not a big stretch to call it by a name. I intend to publicize it and recommend that "PLR" be used as the designation in the next edition of my book, "Mastering Audio." Maybe it will catch on.
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Bob,
100% agree with you.
However - since you are the "man" in some circles regarding this wide topic (I still read your "Mastering Audio" book with purpose :) - it would probably have more weight if you told the TT people to correct it. :)
Just sayin...
VP
I haven't gotten any budge from the TT people. Someone's being very stubborn :-(.
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I haven't gotten any budge from the TT people. Someone's being very stubborn :-(.
Probably hard for them to admit they are wrong.
On another level tho - whilst I totally agree with your technical angle to this - the current "DR" measurement (via TT meter) - right or wrong - does give the layman out there an easy way to quickly determine whether a specific recording has had the crap kicked out of it via dynamic range compression.
Seeing a nice simple number like DR 12 - is much easier to process in one's brain than "6.4 LU" or many of the other values being displayed in MC v19. Looking at the Audio Analysis dialog in v19 - I sometimes feel like I am in an engineering or statistics class :)
When I visit my audio forums and want to gather some VERY quick "intel" on a specific recording prior to purchase - the DR analysis (previously via Foobar or the TT standalone meter and now - thankfully via MC) is really the only weapon out there to give us any indication on how "good" or "bad" the actual "listenability" is on any CD release. If someone has purchased a new CD and has ran it thru the DR meter - and is kind enought to post the results - that's perfect by me.
While I am all about technical "standards"- I do not think that moving to "ITU" and "PLR" and so on - will actually mean a whole lot to those simply wanting to find out if a CD is crushed to death. I agree the naming (or what the TT should be actually measuring) could/should be changed up to be more technically correct...but the pure simplicity of a single number popping out per track is what sells it for me.
At the end of the day - I really don't think a lot of folks actually care what the TT meter measuring - they just want to take comfort in the fact that if they see a 5 on the DR analysis for a specific CD - there's a very high probability that it's a sonic disaster and to stay away from it.
VP
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Vocalpoint...... Yeah, use the TT meter, it gives us something useful. But please tell everyone that you recommend the meter to that it's not measuring "dynamic range", it's measuring a sort of crest factor. And the two terms are very different and should not be confused.
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Yeah, use the TT meter, it gives us something useful. But please tell everyone that you recommend the meter to that it's not measuring "dynamic range", it's measuring a sort of crest factor. And the two terms are very different and should not be confused.
Unfortunately - most folks that I usually yak with regarding the current "DR" meter wouldn't know crest factor from Crest toothpaste :). Most of these folks are simply trying to get thru life trying to find decent music to listen to and not get burned by the current "remaster" myths out there. This little doodad may not be technical correct - but it's easy. That's all these folks really want.
And if no one has mentioned it yet - very nice to see you on this board BTW.
Cheers!
VP
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I'm sorry I got to ask ... what is dynamic range if its not what Dynamic Range (DR) or Dynamic Range (R128) are measuring? I'm confused ...
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I'm sorry I got to ask ... what is dynamic range if its not what Dynamic Range (DR) or Dynamic Range (R128) are measuring? I'm confused ...
Dynamic range is the difference between the loudest and quietest part of the track.
Dynamic Range R128 is a measure of dynamic range, but a statistically weighted one to provide more useful results, that the EBU call "Loudness Range". (LRA)
Dynamic Range (DR) is a measure of Crest Factor, which is not related to Dynamic Range at all - it compares the peak loudness to the average loudness. It's a good measure of how dynamic a track sounds though. (as opposed to sounding "flat" and "compressed")
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I think I get what the 2 measurements are but aparently some people seem strongly opposed to calling DR "dynamic range".
It's probably all lost on me anyways, I'm the kind that can't tell cresh factor from toothpaste ;D, but seriously, isn't it just a matter of interpretation?
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I think I get what the 2 measurements are but aparently some people seem strongly opposed to calling DR "dynamic range".
It's probably all lost on me anyways, I'm the kind that can't tell cresh factor from toothpaste ;D, but seriously, isn't it just a matter of interpretation?
As long as you know what a term means to you, then you can call it "cupcakes" if you like. But for clear communication amongst professionals and serious lay people, it pays for a high quality application like JRiver to use correct, unambiguous terminology.
My two cents,
Bob
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As long as you know what a term means to you, then you can call it "cupcakes" if you like. But for clear communication amongst professionals and serious lay people, it pays for a high quality application like JRiver to use correct, unambiguous terminology.
Thanks, your new rating system is really helping me understand my library.
Unfortunately it just tends to highlight the sorry state of music releases these days. :(
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Thanks, your new rating system is really helping me understand my library.
Unfortunately it just tends to highlight the sorry state of music releases these days. :(
I appreciate the underlying dilemma about terminology, but that screencap is the funniest thing I've seen all week. Not sure why, but it just completely creased me up ;D
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Funny.
As long as you know what a term means to you, then you can call it "cupcakes" if you like. But for clear communication amongst professionals and serious lay people, it pays for a high quality application like JRiver to use correct, unambiguous terminology.
My two cents,
Bob
Of course, and I appreciate that. I didn't mean to imply the opposite. My previous post didn't quite come out the way I intended. I was tired, I rushed it and wanted to go to bed. And the toothpaste joke was just a silly joke, nothing more. I'm actually trying to understand this.
I've read this (http://tech.ebu.ch/docs/tech/tech3342.pdf) before and I just read it again with the wiki page on crest factor (https://en.wikipedia.org/wiki/Crest_factor). I'm honestly trying to understand :P. I went back to the RR DR site (http://productionadvice.co.uk/blog/) and found a link to this (http://www.soundonsound.com/sos/sep11/articles/loudness.htm) which was also a site referenced above.
From that EBU – TECH 3342 pdf I found:
Loudness Range should not be confused with other measures of dynamic range or crest factor, etc.
So there's that. Crest factor is not dynamic range. I get that. Then what is dynamic range (which is actually what I meant to ask all along)?
I found this:
There remains the question of whether one should use such a term as 'dynamic range' at all: there is no official definition for it, and it may be confused with the dynamic range of a recording medium, which is basically the difference between the highest and lowest level it can handle. During the course of this article, therefore, I won't talk about 'dynamic range' in relation to a piece of music. Instead, I will be using 'RMS variability', or more generally 'dynamic variability'. The term 'dynamic range' will be reserved for the measure of signal-to-noise ratio of a recording medium. I will use the term 'loudness range' in strict reference to the EBU 3342 document, and the term 'loudness variability' in other cases involving loudness instead of RMS.
Is that an accurate description of dynamic range then?
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I seem to have killed this thread somehow ... :-\
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for clear communication amongst professionals and serious lay people, it pays for a high quality application like JRiver to use correct, unambiguous terminology.
Hear, hear.
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Here are the specifications (http://www.dynamicrange.de/sites/default/files/DR-Manual-V1_1-English.pdf) for Dynamic Range (DR) implemented by JRiver.
Some mistake? That's a manual for "TT DYNAMIC RANGE METER VST Plugin", not a specification.
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Back to audio analysis now :)
I want to get some feedback (specifically from Matt) on a oddity I am noticing. This past weekend - I complete our whole home move to v19 and everything went great. Of course - it took may hours to analyze the entire library using the new v19 toolkit - but I have noticed an issue - that may be related to a thread I brought up several weeks back where MC was choking on MP3 conversions with 4 files at a time.
Since I had 60000 tracks to analyze over this weekend - I spent a bit of time watch MC do it's thing and noticed a peculiar "lag" in the first thread of a 4 thread analysis cycle. While the first track (Thread) seems to start with the others - it then starts to slowly grind along 5%, 9%, 11% sometimes stopping and waiting for awhile etc etc while the other 3 active rip right through from 0% to 100%....then when the next group of four are targeted - another thread seems to always lagging behind the other three.
Wondering if anyone else has noticed this and if there was an speed issue or not. I realize that all track analysis is unique but I am wondering why that first thread is so slow in completing it's pass while the other three threads seem to fly.
This behavior is consistent on my workstation and can be better seen when doing 30 or 40 at a time...one thread will always be noticeably lagging way behind the other three.
Appreciate any comments from the field...
VP
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Since I had 60000 tracks to analyze over this weekend - I spent a bit of time watch MC do it's thing and noticed a peculiar "lag" in the first thread of a 4 thread analysis cycle. While the first track (Thread) seems to start with the others - it then starts to slowly grind along 5%, 9%, 11% sometimes stopping and waiting for awhile etc etc while the other 3 active rip right through from 0% to 100%....then when the next group of four are targeted - another thread seems to always lagging behind the other three.
This happens all the time - I suspect it's when MC is limited by disk/network speeds.
I have often requested that MC tries to assign a drive to each CPU core rather than working down the list of files sequentially to try and move the limiting factor back to your CPU rather than disk/network access.
I'd also like to see MC be "smarter" about how many files it's analyzing when limited by disk I/O.
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This happens all the time - I suspect it's when MC is limited by disk/network speeds.
But I still see this behavior on local files...certainly not limited by disk I/O...that first thread always stands around like it has no idea what it should be doing :) Over the network - yes - could be anything - but that said - the other three threads - network or local - still burn through with lighting speed...
Something is going on with that first thread...maybe it's normal or by design?
VP
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I've not seen this, i typically use 3 threads and they all run just fine at about the same speed. More threads caused disc IO issues when analyzing video files, so i settled on 3 (it slowed down all analysis tasks, not just one, though)
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I've not seen this, i typically use 3 threads and they all run just fine at about the same speed. More threads caused disc IO issues when analyzing video files, so i settled on 3 (it slowed down all analysis tasks, not just one, though)
I will crank it back to 3 today and see what happens.
VP
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Funny.
Of course, and I appreciate that. I didn't mean to imply the opposite. My previous post didn't quite come out the way I intended. I was tired, I rushed it and wanted to go to bed. And the toothpaste joke was just a silly joke, nothing more. I'm actually trying to understand this.
I've read this (http://tech.ebu.ch/docs/tech/tech3342.pdf) before and I just read it again with the wiki page on crest factor (https://en.wikipedia.org/wiki/Crest_factor). I'm honestly trying to understand :P. I went back to the RR DR site (http://productionadvice.co.uk/blog/) and found a link to this (http://www.soundonsound.com/sos/sep11/articles/loudness.htm) which was also a site referenced above.
From that EBU – TECH 3342 pdf I found:
So there's that. Crest factor is not dynamic range. I get that. Then what is dynamic range (which is actually what I meant to ask all along)?
I found this:
Is that an accurate description of dynamic range then?
There may be no official standard but we know good DR when we hear it! The Loudness war movement is real and correlates well with my taste, R128 seems worthless. Thanks for incorporating DR, I was just starting to think the upgrade was farce, you squeaked her out though.
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There may be no official standard but we know good DR when we hear it! The Loudness war movement is real and correlates well with my taste, R128 seems worthless. Thanks for incorporating DR, I was just starting to think the upgrade was farce, you squeaked her out though.
I'm not sure I understand why you think R128 Leveling seems worthless - it directly counteracts the one of the main problems caused by the loudness war. (tracks being played too loud)
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Hi, having in my playlist a mix of red-book, 24/88, 24/96, 24/192 and DSD (128 bit stream) files, I've noticed that in the new 87 version, the DSD files have a lower volume than before. And that for DSD the volume leveling is not active. I keep my rule to introduce no changes in the signal path, but I was hit by a large swing on a red-book file after several DSD ones.
I'm missing something?
Regards,
Vlad
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If you use DSD Bitstreaming, Volume Leveling would be disabled.
If you have volume leveling active for PCM tracks, and disabled for DSD tracks, you might experience changes in the level.
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I want to copy 30 tracks from my collection to a temp directory and then audio analyse and then volume level all the tracks to each other.
I am playing the tracks via winamp with a broadcast module thru a third party server site to a chat forum that i DJ at (if you will). Therefore I need MC 19 to physically change the volume level data inside each of the 30 tracks, once again so that all tracks are volume leveled to each other when they are played via winamp.
Can MC 19 do what I am requesting?, any help would be appreciated. ( I am using MC 18)
Terry :)
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I have a question about using the data from audio analysis to create playlists.
I have created several "mellow" smartlists based on the following parameters from the old audio analysis:
BPM <= 75 and Intensity <=2
Now that 'Intensity' is gone, all of these smartlists are empty. How do I use the new tools to create similar "mellow" playlists?
I understand that the dynamic range measures the difference between the quietest and loudest levels in a track and that if the dynamic range is small, the track is probably more intense. Tracks that have some quiet passages and some loud passages will have a large dynamic range. This doesn't really help me find "mellow" music. For example, a track of acoustic guitar played quietly will probably have a low dynamic range if there aren't any silent parts.
I think the volume or peak levels would also relate to the mellowness of a song. The volume or peak level of a mellow song should not be as large as the volume or peak level of an intense song. However, the values shown in MC19 don't look like they are absolute values. They look like corrective values. So I am not sure how I could use them in this case.
Can someone help me come up with a formula to create a "mellow" playlist?
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ok, first time using replay gain and auto leveling.
I took a new country album and an old country album with vastly different sound volume levels. I audio analzed the two albums and then during playback used volume leveling with automatic function. The two albums now play at the same level, great it works.
From what I understand the albums now will play at the same volume in another player like winamp, I played them in winamp and sure enough the volume levels are the same, how is this done, winamp doesnt have a volume leveling checkbox to check.
thanks
Terry
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From what I understand the albums now will play at the same volume in another player like winamp, I played them in winamp and sure enough the volume levels are the same, how is this done, winamp doesnt have a volume leveling checkbox to check.
If the player has support for ReplayGain, it will read the tags Media Center has written to the file and use them. (though this "ReplayGain" data is actually derived from the R128 analysis)
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If the player has support for ReplayGain, it will read the tags Media Center has written to the file and use them. (though this "ReplayGain" data is actually derived from the R128 analysis)
If I audio analyze a bunch of mp3's and then format convert them to flac, is the replay gain / volume leveling info still retained?
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If I audio analyze a bunch of mp3's and then format convert them to flac, is the replay gain / volume leveling info still retained?
Out of curiosity, why would you do that?
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I'm starting to feel like I need a PhD in audio production to understand this :P
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Since DR is coming next build (see above), you might wait to analyze.
I'm lost. Did this ever happen?
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19.0.26 (8/13/2013)
3. NEW: Added DR dynamic range analysis to the audio analyzer.
See also:
http://wiki.jriver.com/index.php/File_Properties_%28tags%29#Dynamic_Range_R128 (http://wiki.jriver.com/index.php/File_Properties_%28tags%29#Dynamic_Range_R128)
http://yabb.jriver.com/interact/index.php?topic=82025.0 (http://yabb.jriver.com/interact/index.php?topic=82025.0)
http://yabb.jriver.com/interact/index.php?topic=81980.0 (http://yabb.jriver.com/interact/index.php?topic=81980.0)
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will analyzing audio for 8000 tracks (all flac with cover art 16 bit or higher) take an extremely long time? because every time i open jrmc it is still on track one.
;D
ideas?
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My analysis over under 20k tracks took took most of the night; I don't recall how long exactly, but it was done by morning. The time will also depend on your machine's speed.
The analysis should not keep restarting with track 1 each time. The calculated values are stored in the library. Are your files imported into the main library, or is this some DLAN library from a DLNA server?
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My analysis over under 20k tracks took took most of the night; I don't recall how long exactly, but it was done by morning. The time will also depend on your machine's speed.
The analysis should keep restarting with track 1 each time. The calculated values are stored in the library. Are your files imported into the main library, or is this some DLAN library from a DLNA server?
so after it analyzes them once it has the values it needs, yes? im trying to figure out why, when I open JRiver every single time, it seemingly starts with the same "cache" of files(same count, etc) and takes forever and a month of sundays to do them...... (ie seemingly several days) would a log help, i mean could yall tell me if everything is hunky dory
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I corrected my typo above. Keyword NOT was missing.
Are you able to tag files and have the values remain?
Please describe how you import files, any import settings you've configured, and where the files are located.
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yes, I can tag files, they are fine.
I get cds so frequently(or vinyl rips) that I am constantly using the "run auto import" feature(so the new stuff will show up.
the bulk of the audio /video files IS on an external 2tb disc. program files are on the C drive.
two folders on the external drive are in the library, and two folders in the system(the system pictures folder, and also a temporary location I name "limbo" where I keep downloaded stuff from HD tracks that ive bought(if I dont like it it doesnt go to the main library on the external drive)
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When this feature first rolled out I created a smartlist that showed files that were as yet not analyzed. The file count most recently was negligible and limited to files recently imported. With the release of .116 my count is now up to some 46,000 files (of approx. 377,000) needing analysis. The files that need reanalysis are without exception audio files. Any ideas?
Smartlist includes Volume Level (R128) is empty and Dynamic Range (R128) is empty.
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Thanks guys for pointing out that the two files were on the same album. That had flown right over my head, and I'm sure that's the explanation.
If they're different albums, I'd recommend giving them different names. We might also be able to leverage the album analyzer which considers file path, but this opens up a new set of problems with users that don't reliably use album folders.
Yes please. It would be great to be able to configure what is defined as an album. I've found that if I give the album a different name it doesn't get cover art correctly and if I give it the correct name then MC has problems with differentiating during audio analysis.
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i still have no idea how to ID which files in my library are giving jriver audio analysis problems.
if there is a way to filter them, or some app for that purpose, that would be fantastic. would prefer to do it with jriver tho
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i still have no idea how to ID which files in my library are giving jriver audio analysis problems.
if there is a way to filter them, or some app for that purpose, that would be fantastic. would prefer to do it with jriver tho
If you manually run analysis, it will either error out, or stop progressing on bad files.
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I'm trying this feature for the first time, and so far I have analyzed about 20000 of my ca 90000 files.
I have added the relevant columns like "volume level (replay gain)" etc. to see whats happening during the process. Its very interesting.
But, so far it has only analyzed my normal flac's and ape's with new values, none of my 24/48, 24/96, 24/192 or my sacd iso's have been analyzed or show any values in the new columns..?!? Why is that..?!?
Am I missing something..?!?
Edit.
A few of my 24/48 and 24/96 does get analyzed. But the majority wont...?!?
Edit no 2 8).
Manual analyze works! That makes me happy..!! I see now that analyzing sacd iso's takes a long time, at least on my Core2 server. So maybe this issue was a non-issue. Sorry.. ;D
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This expression will let you see how much headroom your tracks require. (it could probably be cleaned up - but it works)
Delimit(if(isempty([Peak Level (R128)]),,formatnumber(math(removecharacters(left([Peak Level (R128)],5),/ /+,0)+RemoveCharacters([Volume Level (R128)],/ LU,0)+1),1)),/ dB,)
Sorry for resuscitating an older thread, but I have a question regarding this expression and Volume Levelling in general. I have analysed audio in my library, and if I understand this expression it adds Peak Level R128 to Volume Level R128 plus 1. I have a track on Grofe Grand Canyon Suite album, track "Crickets and Distant Thunder". It shows:
Volume Level R128: 12.9 LU
Dynamic Range R128: 26 LU
Peak Level R128: -5.1 dBTP, -5.1 Left, -14.6 Right
Dynamic Range: 12
Headroom (from above expression): 8.8 dB
If I understand it correctly, this means it should clip heavily if it is volume levelled at the analysed setting of 12.9. My question is why Mac MC 20.0.40 (oops, should I start a thread in the new forum, Mac or Win?) calculate this value at a level that would lead to such high amount of clipping? Also, why is the Dynamic Range number consistently different from the Dynamic Range R128?
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Sorry for resuscitating an older thread, but I have a question regarding this expression and Volume Levelling in general. I have analysed audio in my library, and if I understand this expression it adds Peak Level R128 to Volume Level R128 plus 1. I have a track on Grofe Grand Canyon Suite album, track "Crickets and Distant Thunder". It shows:
Volume Level R128: 12.9 LU
Dynamic Range R128: 26 LU
Peak Level R128: -5.1 dBTP, -5.1 Left, -14.6 Right
Dynamic Range: 12
Headroom (from above expression): 8.8 dB
If I understand it correctly, this means it should clip heavily if it is volume levelled at the analysed setting of 12.9. My question is why Mac MC 20.0.40 (oops, should I start a thread in the new forum, Mac or Win?) calculate this value at a level that would lead to such high amount of clipping? Also, why is the Dynamic Range number consistently different from the Dynamic Range R128?
You have it backwards; the 12.9 number is the number of dB by which the track is attenuated for volume leveling, not boosted. Volume leveling only works by attenuation, it never rarely adds boost [EDITED: because I was incorrect).
The two dynamic ranges are different because they measure very different things. I explained the difference between the two in detail here: http://yabb.jriver.com/interact/index.php?topic=84489.msg577389#msg577389
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You have it backwards; the 12.9 number is the number of dB by which the track is attenuated for volume levelling, not boosted. Volume levelling only works by attenuation, it never adds boost.
If I understand you correctly, this might then be a bug, as in this case the Volume Level R128 is, actually, a positive figure of 12.9, not a negative number, like for majority of my tracks. I understood that negative meant attenuation but positive values meant volume boosting. I have 332 tracks for which Volume Level is a positive number. Why would this happen?
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If I understand you correctly, this might then be a bug, as in this case the Volume Level R128 is, actually, a positive figure of 12.9, not a negative number, like for majority of my tracks. I understood that negative meant attenuation but positive values meant volume boosting. I have 332 tracks for which Volume Level is a positive number. Why would this happen?
I've never seen a track that required a positive adjustment of 12 dB, so I misunderstood what you posted; my apologies.
Volume levelling is trying to achieve a specific level (83 dB at -20dBFS) and some tracks will have positive values because the tracks in question are very, very quiet tracks. In order to make them "the same volume" as other tracks they would need to have volume added to them. I misspoke before when I said volume leveling never boosts; in practice volume leveling very rarely adds boost because the vast majority of tracks have a peak level at or near 0dBFS, so no volume can be safely added. Your track is unusual in that it has a very low peak level as well, so about 4 dB could be safely added, but that's it.
So the track would need 12 dB of boost to sound "the same" volume as other tracks, but because it peaks at -5.1dB, only 4 dB can be safely added, and that's the most volume leveling would ever add. Sorry for the miscommunication/information!
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Your track is unusual in that it has a very low peak level as well, so about 4 dB could be safely added, but that's it.
So the track would need 12 dB of boost to sound "the same" volume as other tracks, but because it peaks at -5.1dB, only 4 dB can be safely added, and that's the most volume leveling would ever add. Sorry for the miscommunication/information!
Many thanks for your explanations, they are very helpful. Let me check if I still understand it the way you have explained, though: in this track, even though the Volume Level R128 has been calculated as +12.9 LU, but because the Peak Level R128 is -5.1 dBTP, Volume Levelling would not add the 12.9 boost, but only the smaller amount, that is of 5.1 dB, in order to prevent clipping. Is that correct or have I still misunderstood? Many thanks for your patience...
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Many thanks for your explanations, they are very helpful. Let me check if I still understand it the way you have explained, though: in this track, even though the Volume Level R128 has been calculated as +12.9 LU, but because the Peak Level R128 is -5.1 dBTP, Volume Levelling would not add the 12.9 boost, but only the smaller amount, that is of 5.1 dB, in order to prevent clipping. Is that correct or have I still misunderstood? Many thanks for your patience...
You're very close, and you have the principle correct, but with one tweak: volume leveling typically leaves a 1dB "margin of error" to make absolutely sure no clipping occurs. So the most I would expect volume leveling would add is 5.1 - 1 = 4.1dB
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While aware of its impact audibly, definitely a newbie to the world of measured DR!
Recently found out about the <dr.loudness-war.info/> database and, after googling a bit on using Audacity to generate DR values, learned that MC will generate them. Very nice, BTW! Not sure how comparable the results might be but I am sure they are close enough for my purposes since I am not doing any automatic Volume Leveling. Information in the Wiki on Analyze Audio was a bit lean. It would be helpful for us that are less technically savvy to have a (simple?) explanation of what the various results are and/or mean to the average listener. Skimming this thread made my head hurt! :-)
I see where I can select track(s) of an album, etc., and then activate the Analyze Audio option. Is there some way to select ALL tracks in my library and generate the data "enmasse"? IF so and I chose to analyze the full database, then I guess I would need to keep track of what was added subsequently and batch analyze?!
Can the Analyze Audio be used on PCM formats other than 16/44? How about DSD?
Thanks in advance.
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While aware of its impact audibly, definitely a newbie to the world of measured DR!
I see where I can select track(s) of an album, etc., and then activate the Analyze Audio option. Is there some way to select ALL tracks in my library and generate the data "enmasse"? IF so and I chose to analyze the full database, then I guess I would need to keep track of what was added subsequently and batch analyze?!
Can the Analyze Audio be used on PCM formats other than 16/44? How about DSD?
Thanks in advance.
go to your audio view, select all the tracks (shift-click), and analyze the audio. if you have a lot of tracks it will take some time!
in the analyze audio menu, there is a checkbox "skip analyzed files". check this, and never think about which files have been done or not, as MC will skip previously analyzed files
you can analyze any PCM bit/sample rate, and DSD .dff files. you can also analyze the audio for video files using the same tool
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go to your audio view, select all the tracks (shift-click), and analyze the audio. if you have a lot of tracks it will take some time!
in the analyze audio menu, there is a checkbox "skip analyzed files". check this, and never think about which files have been done or not, as MC will skip previously analyzed files
you can analyze any PCM bit/sample rate, and DSD .dff files. you can also analyze the audio for video files using the same tool
Thank you, ferday.
From info you provided, figured out that I had to open up the AUDIO tree to get to the folder where the source files are stored, double click on folder to show all files, do CTRL A to select all files, and then ANALYZE AUDIO. The "skip analyzed files" was already checked (default?)
Does it matter whether you select "1 file at a time", "2 files at time", etc.?
Tried googling on the ANALYZE AUDIO data column headings but not finding a relatively understandable (to me) description of what they mean and their implication on listening. Again, it would be great if the Wiki were expanded to include this information.
I am using MC at probably close to its simplest implementation but continue to be impressed with its capabilities! :D
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Just found the TAGS section of the Wiki which has a glossary of all terms including those in the column headers of the ANALYZE AUDIO data screen.
Given site thoroughness with most things, figured that there WAS a description of terms somewhere but did not associate the ANALYZE AUDIO with the TAGS function!
Thanks.
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MC19 makes several important improvements to audio analysis and volume leveling:
- Adoption of the R128 industry standard (http://tech.ebu.ch/loudness) to analyze the loudness and dynamic range of content
- Ability to analyze audio for video files, including surround sound
- Smarter Volume Leveling that automatically respects intentional between track levels when playing from an album
- Volume Leveling works together with Adaptive Volume's peak level normalization (http://yabb.jriver.com/interact/index.php?topic=82023.0)
- Peak level is reported in decibels, measured as an R128 compliant True Peak, and reported per channel
This is a big topic, so I'll try to revisit it and post more details about the changes and the motivation at a later date.
Regarding this not uncommon problem https://benchmarkmedia.com/blogs/application_notes/intersample-overs-in-cd-recordings?_pos=1&_sid=0eeb1f150&_ss=r
https://gearspace.com/board/mastering-forum/1401406-intersample-clipping-audible.html
I had recently asked about it here. https://yabb.jriver.com/interact/index.php?topic=135725.0 , But it's still not clear to me, which if any player improvements allow correction of this specific problem. If yes, how can JRiver be set up to reduce or eliminate this kind of distortion?