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  1. A couple of minor modifications to the Now Playing tab. The Title line only displays the title when there's title info in tags. Previously it displayed the file name in the absence of title info, but there's a "File" line already displaying the file name.

    I moved the file type (mp3/aac etc) from the "File" line to the "Codec" line. I think it looks a bit neater.

    I can't decide if I prefer it, but here's a new Playlist Title column too. It no longer appends "file name" when it's displaying the file name. Instead, if a file name is being displayed because there's no Title info in tags, it's displayed in brackets.

    Code:
    Title  ( file name ) - Aligned Left
    $puts(A,$ifequal($meta_num(title),0,'(' %filename% ')',%title%))[$if(%isplaying%,>>>$get(A)<<<,$get(A))]
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    Last edited by hello_hello; 20th Nov 2018 at 06:07.
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  2. Originally Posted by hello_hello View Post
    A couple of minor modifications to the Now Playing tab. The Title line only displays the title when there's title info in tags. Previously it displayed the file name in the absence of title info, but there's a "File" line already displaying the file name.

    I moved the file type (mp3/aac etc) from the "File" line to the "Codec" line. I think it looks a bit neater.

    I can't decide if I prefer it, but here's a new Playlist Title column too. It no longer appends "file name" when it's displaying the file name. Instead, if a file name is being displayed because there's no Title info in tags, it's displayed in brackets.

    Code:
    Title  ( file name ) - Aligned Left
    $puts(A,$ifequal($meta_num(title),0,'(' %filename% ')',%title%))[$if(%isplaying%,>>>$get(A)<<<,$get(A))]

    Ah nice one, I used your previous one but I couldn't figure out how to get the default columns back lol, this foobar is complicated for some parts but I do like it, I have moved away from windows media player now lol plus how did you get the graphic equalizer bars going at the right hand side?
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  3. Are you referring to the Peak Meter? It's the Peak Meter on the right.

    I added the Splitter Element under each tab of the Tabs Element, then added various elements on the left side, and always the Peak Meter on the right.
    I like having the meter there to show me the output, but I think it's a bit fugly, and I don't want it to be intrusive, which is why I have a tab called "Meter". It's the same as the "Now Playing" tab, but with the Peak Meter large enough for the channel names at the bottom to be readable. I was sick of resizing the Peak Meter, so now I switch tabs instead.

    Actually.... my "Meter" tab is almost the same as my "Now Playing" tab, but above the Peak Meter is the R128 Meter displaying the text. It's also a bit fugly.

    FYI, If you right click on the Peak Meter element, you can run it in full screen mode.
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    Last edited by hello_hello; 20th Nov 2018 at 10:34.
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  4. Been messing about with foobar big time tonight, I have my interface the way I want it, excellent program, need to sort out font colors and stuff but amazed how fast I'm learning it, usually I wouldn't be bothered with so many things to edit and customize but deffo love it, would be good if there was an actual column out there so that you could see if a track is going to say track will be clipped during track gain alteration, but then again there is that option apply gain and prevent clipping but would be good to see it as mp3 gain does, it analyzes the track and afterward it tells you if it would be applying clipping, anyway tell me what you think, need to make some adjustments but with your now playing text, it helps a lot
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    Last edited by circulationds; 20th Nov 2018 at 19:18.
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  5. Your setup is looking pretty good, although the Peak Meter displaying so many channels in the 1st screenshot confuses me. The playing file appears to be stereo. Were you using a DSP to upmix on playback, or something similar?

    I know what you mean about determining if there'll be clipping after adjusting. I generally just apply the adjustment and then check for peaks above 0dB and for those, adjust them down if need be (I don't worry about peaks just a fraction over maximum (up to around 0.5dB). Originally, I added the "Selection Properties" element to the GUI and enabled the display of ReplayGain info, because when multiple files are selected, it stops displaying any TrackGain info and only shows the highest peak for the selected files (as "total peak"). If it's greater than 1.0000, then at least one file has a peak above maximum and I'd know to look for the culprit(s). For a long time, that's how I made checking the peaks after adjusting a bit easier, but now we have the Track Peak info displaying as a Playlist column it's not too hard to check each file after adjusting. Anyway... the 1st screenshot below shows my "File Info" tab with the "selection properties" element displaying the loudest peak, although that tab is a bit redundant now. I should remove it.

    Have you played around with the Facets library viewer yet? Once you start managing your song library you'll probably want to. The two columns on the left of my fb2k setup are Facets columns. If you don't have it added to the GUI as an element, you can still use it via the Library/Facets menu. By default, it opens up something like the second screenshot below. The columns interact, so for me clicking on an Artist name in the left column causes the Title column to display just the tracks by that artist (as per the 3rd screenshot), but that's just how I prefer it. It's quite configurable. fb2k's Library Viewer/Playlist combination of doing things can take a little getting used to, but now I can't imagine doing it any other way.

    Facets has a filter that can be used to effectively divide your media library into sections if you wish to. For example, I have original CD rips in a different location to the MP3 versions. To display them as individual media libraries I've created filters that only show the contents of certain folders (and their sub-folders). It's easy as you don't have to specify the full folder path, just a section of it that's unique, so I have two Filters (configured under Media Library/Facets/Filter). One filter's syntax is something like "My Stuff/MP3s" and the second is "My Stuff/CD Rips". The Filter drop down menu makes it easy to switch between them.

    A library viewer is the way you'd normally add stuff from your media library to a playlist, because you can't play anything unless it's in a playlist. By default, Facets switches between being a Library viewer and a Playlist viewer when you select a playlist tab. I don't like that behaviour so I have it configured to always display the contents of my media library. That way, the left side of my fb2k configuration just displays the media library contents and the right side just the playlist contents. My brain seems to prefer it that way.
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    Last edited by hello_hello; 21st Nov 2018 at 02:03.
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  6. Originally Posted by hello_hello View Post
    Your setup is looking pretty good, although the Peak Meter displaying so many channels in the 1st screenshot confuses me. The playing file appears to be stereo. Were you using a DSP to upmix on playback, or something similar?

    I know what you mean about determining if there'll be clipping after adjusting. I generally just apply the adjustment and then check for peaks above 0dB and for those, adjust them down if need be (I don't worry about peaks just a fraction over maximum (up to around 0.5dB). Originally, I added the "Selection Properties" element to the GUI and enabled the display of ReplayGain info, because when multiple files are selected, it stops displaying any TrackGain info and only shows the highest peak for the selected files (as "total peak"). If it's greater than 1.0000, then at least one file has a peak above maximum and I'd know to look for the culprit(s). For a long time, that's how I made checking the peaks after adjusting a bit easier, but now we have the Track Peak info displaying as a Playlist column it's not too hard to check each file after adjusting. Anyway... the 1st screenshot below shows my "File Info" tab with the "selection properties" element displaying the loudest peak, although that tab is a bit redundant now. I should remove it.

    Have you played around with the Facets library viewer yet? Once you start managing your song library you'll probably want to. The two columns on the left of my fb2k setup are Facets columns. If you don't have it added to the GUI as an element, you can still use it via the Library/Facets menu. By default, it opens up something like the second screenshot below. The columns interact, so for me clicking on an Artist name in the left column causes the Title column to display just the tracks by that artist (as per the 3rd screenshot), but that's just how I prefer it. It's quite configurable. fb2k's Library Viewer/Playlist combination of doing things can take a little getting used to, but now I can't imagine doing it any other way.

    Facets has a filter that can be used to effectively divide your media library into sections if you wish to. For example, I have original CD rips in a different location to the MP3 versions. To display them as individual media libraries I've created filters that only show the contents of certain folders (and their sub-folders). It's easy as you don't have to specify the full folder path, just a section of it that's unique, so I have two Filters (configured under Media Library/Facets/Filter). One filter's syntax is something like "My Stuff/MP3s" and the second is "My Stuff/CD Rips". The Filter drop-down menu makes it easy to switch between them.

    A library viewer is the way you'd normally add stuff from your media library to a playlist because you can't play anything unless it's in a playlist. By default, Facets switches between being a Library viewer and a Playlist viewer when you select a playlist tab. I don't like that behavior so I have it configured to always display the contents of my media library. That way, the left side of my fb2k configuration just displays the media library contents and the right side just the playlist contents. My brain seems to prefer it that way.

    Not played around with the facets yet! I will have a look into that at some point as I want to add my now that's what I call music collection to foobar and organize them albums, with the first screenshot that was a graphic equalizer instead of the peak meter, I selected the wrong one when uploading the images to show you, the files playing where my converted movie audio files, I just wanted to see the ones that where over the 1.0000 peaks. I checked the files, they seem to be ok, and I can't notice clipping in them, to be honest. and the peak meter gain when the movie was at high volume didn't show on the peak meter above 0db, it was just below so I know that there is some give in some files. I will have to look into that more but I have been using your matrix mixer to encode 5.1 to 2 channel AAC then after i do that, i scan them files add the tags then I apply 70DB to alter file volume with lower adjustments to prevent clipping according to peak information, then apply that to file content then i use the final normalization with the rocksteady plugin with the following settings.

    Amplify -3
    Full Amplification 15%
    Maximum Amplification 10DB
    Amp Gain Time 10MS
    Use Smart Limiting


    the reason why I am using the rocksteady after converting just the 5.1 to 2 channel using your matrix settings, there was a couple of files that I couldn't hear the dialogue all that well after adjusting the final volume, so I used the rocksteady to convert to final conversion and it's fine, I had to do all the 5.1 audio files again because I was on peaks at 2.0000 which where far too high and I did notice the clipping when watching films the other night. I have re-encoded them files and checked them, some have peaks just below 1.0000 and slightly over 1.0000 but when playing loud parts of the files and viewing the peak meter there not going past the 0DB mark and they sound alot better than they did before.
    Last edited by circulationds; 21st Nov 2018 at 05:30.
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  7. Are you converting to AAC, then scanning with ReplayGain and converting again to compress? If there's two conversions in your process the first one should ideally be to a lossless format, but I'm not sure I'm understanding correctly.

    To confirm... you have been downmixing and compressing at the same time, and in that order? As per the 1st screenshot below? The audio is still being downmixed before it's compressed but as a single process. The result should be the same as doing it in two stages. If you want to adjust the volume after compressing you can downmix, compress and convert to a lossless format, or you can convert directly to AAC and adjust their volumes losslessly, but to me it doesn't make sense to adjust the volume with ReplayGain before compressing as the compression will change it.

    Unless.... you're adjusting the volume to 70dB(?) before compressing because you think the Rock Steady plugin works better that way. If so you can still do it in a single step (after scanning the source files and saving the Track Gain data). The ReplayGain processing in the second screenshot would adjust the volume to 70dB first, followed by downmixing, and then compression.

    For the record, if you scan 5.1ch audio to determine the volume, it doesn't translate directly to the volume of a stereo version after downmixing. I wish there was a way to scan 5.1ch audio and use the ReplayGain info to adjust the volume by the correct amount while downmixing to achieve a particular target volume, but unfortunately it doesn't work that way. For the purpose of using the ReplayGain data to adjust the volume before compressing though, the audio would be adjusted to the same volume, so the compressor would have a similar amount of headroom to play with, which I assume is the main reason adjusting the volume before compressing.

    Personally, I'd just reduce the volume by a fixed amount, and then there'd be no need to run a Replay Gain scan before downmixig. As per the 3rd screenshot.
    I'd downmix with the Matrix Mixer's "normalise" option enabled, followed by the Amplify DSP to reduce the volume by a fixed amount, followed by the compression. To prevent the Winamp Bridge from limiting any peaks before it converts the audio to 16 bit, the loudest peak should be -6dB or less. Generally the volume for 5.1ch audio should be somewhere in the vicinity of -23 LUFS (around 83dB) so a fixed reduction of 10dB should be plenty, and it should keep the peaks safe from the Winamp Bridge's limiter. How much you reduce the volume is up to you, but I'd be trying to do it all in a single step as much as possible. There's no reason why the result should be any different to doing it in two or more steps, assuming the same volume adjustments are applied etc.
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    Last edited by hello_hello; 21st Nov 2018 at 08:26.
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  8. Originally Posted by hello_hello View Post
    Are you converting to AAC, then scanning with ReplayGain and converting again to compress? If there's two conversions in your process the first one should ideally be to a lossless format, but I'm not sure I'm understanding correctly.

    To confirm... you have been downmixing and compressing at the same time, and in that order? As per the 1st screenshot below? The audio is still being downmixed before it's compressed but as a single process. The result should be the same as doing it in two stages. If you want to adjust the volume after compressing you can downmix, compress and convert to a lossless format, or you can convert directly to AAC and adjust their volumes losslessly, but to me it doesn't make sense to adjust the volume with ReplayGain before compressing as the compression will change it.

    Unless.... you're adjusting the volume to 70dB(?) before compressing because you think the Rock Steady plugin works better that way. If so you can still do it in a single step (after scanning the source files and saving the Track Gain data). The ReplayGain processing in the second screenshot would adjust the volume to 70dB first, followed by downmixing, and then compression.

    For the record, if you scan 5.1ch audio to determine the volume, it doesn't translate directly to the volume of a stereo version after downmixing. I wish there was a way to scan 5.1ch audio and use the ReplayGain info to adjust the volume by the correct amount while downmixing to achieve a particular target volume, but unfortunately it doesn't work that way. For the purpose of using the ReplayGain data to adjust the volume before compressing though, the audio would be adjusted to the same volume, so the compressor would have a similar amount of headroom to play with, which I assume is the main reason adjusting the volume before compressing.

    Personally, I'd just reduce the volume by a fixed amount, and then there'd be no need to run a Replay Gain scan before downmixig. As per the 3rd screenshot.
    I'd downmix with the Matrix Mixer's "normalise" option enabled, followed by the Amplify DSP to reduce the volume by a fixed amount, followed by the compression. To prevent the Winamp Bridge from limiting any peaks before it converts the audio to 16 bit, the loudest peak should be -6dB or less. Generally the volume for 5.1ch audio should be somewhere in the vicinity of -23 LUFS (around 83dB) so a fixed reduction of 10dB should be plenty, and it should keep the peaks safe from the Winamp Bridge's limiter. How much you reduce the volume is up to you, but I'd be trying to do it all in a single step as much as possible. There's no reason why the result should be any different to doing it in two or more steps, assuming the same volume adjustments are applied etc.

    My steps are

    convert to Apple AAC lossless file and use nothing but your matrix to downmix to stereo, once they're complete I scan them converted files and apply a gain of 70DB apply gain without clipping, then I convert them files again, ACC lossless using the Winamp bridge with the settings I mentioned above then scan them and apply a gain of 83db without clipping. I don't scan the 5.1 audio before converting with the matrix mixer, I will do your steps for the next files and see how I get on, but I had to reduce some of the bridge settings as stated above as I was experiencing clipping that's why I did them again, after files have been fully completed I do a scan again and any that are going over peak I reduce the overall volume. i will try your step for the next batch and see how they get on, all in there is only about 120 5.1 channel files so it's not a big deal and not as much time to do it as when i was doing it with pazerra followed by box4 the image below is the completed files all adjusted to 83 db and with apply gain without clipping, they seem to be ok but like i said i had to adjust winamps bridge settings to get these results. hope this makes sense i have added one of the completed conversions so you can listen and tell me if it's ok
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    Last edited by circulationds; 21st Nov 2018 at 10:07.
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  9. circulationds,

    I ran some more tests myself, starting with some 5.1ch audio from a TV show, and found downmixing followed by a -6dB gain reduction, followed by my usual RockSteady settings worked quite well, resulting in peaks of around -2 dB to -3 dB. When I tried some loud audio from a couple of movies it was a different story though. One was 5.1ch DTS and the other 5.1ch AC3. I scanned them just to see, and the DTS audio had a volume of -17.62 LUFS with a Track Peak of -00.17 dB. For the AC3 audio the volume was -13.90 LUFS with a True peak of +07.03 dB. Chances are the AC3 contained info to reduce the volume on playback, but I have the AC3 decoder configured to ignore it. That's my theory anyway. I might check it tomorrow.

    Anyway, I experimented a bit, trying to get away with not having to reduce the volume much, and without reducing the compression if possible. After a few trial and error runs, the following seemed to do the trick.
    First the matrix mixer in the DSP chain with my usual settings and "normalise" checked.
    Second the Amplify DSP set to -12 dB.
    Third the WinAmp Bridge with my usual Rock Steady settings, with the exception of lowering the "Amplify up to" setting to -6dB.

    AmplifyTo -6
    FullAmpTo 30
    MaxAmpDB 15
    AmpGainTime 10
    SmartLimit yes
    JointStereo yes
    WindowWidth 75
    RelPos 100
    CalcRmsPer 32

    The end result for the stereo output was a volume of -25.04 LUFS and a Track Peak of -00.61 dB for the DST source, and for the AC3 it was -23.12 LUFS and a Track Peak of -00.87 dB.
    They both only just came in under maximum for the peak values, but I suspect the two sources I was playing with would be pretty close to worse case scenario, so I think those settings would be fine most of the time. At worst, the volume should only need to be reduced by a few dB more.

    I've just downloaded your sample but it's pretty late here now so I'll listen tomorrow and report back.

    Cheers.
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  10. Originally Posted by hello_hello View Post
    circulationds,

    I ran some more tests myself, starting with some 5.1ch audio from a TV show, and found downmixing followed by a -6dB gain reduction, followed by my usual RockSteady settings worked quite well, resulting in peaks of around -2 dB to -3 dB. When I tried some loud audio from a couple of movies it was a different story though. One was 5.1ch DTS and the other 5.1ch AC3. I scanned them just to see, and the DTS audio had a volume of -17.62 LUFS with a Track Peak of -00.17 dB. For the AC3 audio the volume was -13.90 LUFS with a True peak of +07.03 dB. Chances are the AC3 contained info to reduce the volume on playback, but I have the AC3 decoder configured to ignore it. That's my theory anyway. I might check it tomorrow.

    Anyway, I experimented a bit, trying to get away with not having to reduce the volume much, and without reducing the compression if possible. After a few trial and error runs, the following seemed to do the trick.
    First the matrix mixer in the DSP chain with my usual settings and "normalise" checked.
    Second the Amplify DSP set to -12 dB.
    Third the WinAmp Bridge with my usual Rock Steady settings, with the exception of lowering the "Amplify up to" setting to -6dB.

    AmplifyTo -6
    FullAmpTo 30
    MaxAmpDB 15
    AmpGainTime 10
    SmartLimit yes
    JointStereo yes
    WindowWidth 75
    RelPos 100
    CalcRmsPer 32

    The end result for the stereo output was a volume of -25.04 LUFS and a Track Peak of -00.61 dB for the DST source, and for the AC3 it was -23.12 LUFS and a Track Peak of -00.87 dB.
    They both only just came in under maximum for the peak values, but I suspect the two sources I was playing with would be pretty close to worse case scenario, so I think those settings would be fine most of the time. At worst, the volume should only need to be reduced by a few dB more.

    I've just downloaded your sample but it's pretty late here now so I'll listen tomorrow and report back.

    Cheers.

    No worries, I have checked some of my converts and they seem to be fine with the new settings I posted, checked that 10 Cloverfield compared to the other one and it's definitely better than the previous one, I know you don't understand why I'm doing it in a two-step process but I have to make sure the files are ok as I want to delete the originals but keep the extracted original audio just in case after I have a satisfying result with the converted movies as I don't see the point in keeping the originals if I get the converts to a satisfactory result. I did try your method with the downmix to 5.1 then use the amplify and then rocksteady in the Dsp line and I was getting the same results, but it's just me I want to make sure, but either way I'm enjoying using foobar anyway so it's not a big deal to me if I redo the converts i just like to perfect the work i do on them, bit of OCD lol
    Last edited by circulationds; 22nd Nov 2018 at 08:40.
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  11. Originally Posted by circulationds View Post
    Originally Posted by hello_hello View Post
    circulationds,

    I ran some more tests myself, starting with some 5.1ch audio from a TV show, and found downmixing followed by a -6dB gain reduction, followed by my usual RockSteady settings worked quite well, resulting in peaks of around -2 dB to -3 dB. When I tried some loud audio from a couple of movies it was a different story though. One was 5.1ch DTS and the other 5.1ch AC3. I scanned them just to see, and the DTS audio had a volume of -17.62 LUFS with a Track Peak of -00.17 dB. For the AC3 audio the volume was -13.90 LUFS with a True peak of +07.03 dB. Chances are the AC3 contained info to reduce the volume on playback, but I have the AC3 decoder configured to ignore it. That's my theory anyway. I might check it tomorrow.

    Anyway, I experimented a bit, trying to get away with not having to reduce the volume much, and without reducing the compression if possible. After a few trial and error runs, the following seemed to do the trick.
    First the matrix mixer in the DSP chain with my usual settings and "normalise" checked.
    Second the Amplify DSP set to -12 dB.
    Third the WinAmp Bridge with my usual Rock Steady settings, with the exception of lowering the "Amplify up to" setting to -6dB.

    AmplifyTo -6
    FullAmpTo 30
    MaxAmpDB 15
    AmpGainTime 10
    SmartLimit yes
    JointStereo yes
    WindowWidth 75
    RelPos 100
    CalcRmsPer 32

    The end result for the stereo output was a volume of -25.04 LUFS and a Track Peak of -00.61 dB for the DST source, and for the AC3 it was -23.12 LUFS and a Track Peak of -00.87 dB.
    They both only just came in under maximum for the peak values, but I suspect the two sources I was playing with would be pretty close to worse case scenario, so I think those settings would be fine most of the time. At worst, the volume should only need to be reduced by a few dB more.

    I've just downloaded your sample but it's pretty late here now so I'll listen tomorrow and report back.

    Cheers.

    No worries, I have checked some of my converts and they seem to be fine with the new settings I posted, checked that 10 Cloverfield compared to the other one and it's definitely better than the previous one, I know you don't understand why I'm doing it in a two-step process but I have to make sure the files are ok as I want to delete the originals but keep the extracted original audio just in case after I have a satisfying result with the converted movies as I don't see the point in keeping the originals if I get the converts to a satisfactory result. I did try your method with the downmix to 5.1 then use the amplify and then rocksteady in the Dsp line and I was getting the same results, but it's just me I want to make sure, but either way I'm enjoying using foobar anyway so it's not a big deal to me if I redo the converts I just like to perfect the work I do on them, bit of OCD lol

    Just an update, it has come to my attention, that some movie files have already been tampered with before I have tried to normalize them, I watched my converted scary movie 5 and its track volume is -24.37 LUFS and it's peak is 0.385749 well in range as you did mention a 0.5 maximum peak is ok but in certain scenes there was a big massive distortion issue in the movie. I scanned the original file, it is coming up at a "wapping" -10.82 LUFS and a peak value of 1.835156 so now I know that I'm not making any mistakes it has been the original files that are too high and there's nothing I can do to fix them apart from getting replacement originals. I have opened the files with mktoolnix as I thought maybe there were tags in them and could remove them but nothing, least I know now that when I was trying your settings with the rocksteady plugin, no matter how much I adjusted them, it wouldn't fix the issue as the originals have already been clipped, so there for ruined by whoever done the editing on them. Going to try audacity as I know that has a clipping fix on it, worth a try anyway.


    Just fixed the scary movie 5 file with clip fix in audacity, so that's good to know I can fix original files are clipping, there is only a few.
    Last edited by circulationds; 23rd Nov 2018 at 11:20.
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  12. I remember saying I tend not to fuss too much over peaks just a tad above maximum. +0.5 dB would probably be the most I'd live with. Maybe a tad more if I was in a good mood at the time.
    I think that's roughly 1.06000 as a percentage. Is that what you meant by a maximum peak of 0.5?

    The file you're scanning with peaks above maximum..... don't forget some lossy formats do store peaks above 0dB without actually clipping them. Are you sure the peaks are physically clipped rather than being clipped on playback? Do they looked clipped in Audacity if you lower the volume? Audacity will probably import the audio as 32 bit float so I don't think the peaks would be clipped while it's being imported as such, although I'm not 100% what Audacity does.
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  13. Originally Posted by hello_hello View Post
    I remember saying I tend not to fuss too much over peaks just a tad above maximum. +0.5 dB would probably be the most I'd live with. Maybe a tad more if I was in a good mood at the time.
    I think that's roughly 1.06000 as a percentage. Is that what you meant by a maximum peak of 0.5?

    The file you're scanning with peaks above maximum..... don't forget some lossy formats do store peaks above 0dB without actually clipping them. Are you sure the peaks are physically clipped rather than being clipped on playback? Do they look clipped in Audacity if you lower the volume? Audacity will probably import the audio as 32 bit float so I don't think the peaks would be clipped while it's being imported as such, although I'm not 100% what Audacity does.
    Yeah that's what I mean a maximum peak of 0.5, I remember you saying that lossy can store peaks above 0DB but when i played the movie both on tv and pc the same scenes where distorted really bad, like over volume, I played my normalized audio file as well and the same was happening on tv and pc

    yeah they do look clipped in audacity, there are only about 20 files that need working, will finish them off today then all the movies are done, when i open the audio file in audacity you can tell that they are clipped as you do a scan for clipped sections and then use fix clip, that's what i did, will only use audacity for any i come across in the future as it is time-consuming and with foobar i just set my converts up before bed and let them do there thing.
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  14. Originally Posted by circulationds View Post
    Originally Posted by hello_hello View Post
    I remember saying I tend not to fuss too much over peaks just a tad above maximum. +0.5 dB would probably be the most I'd live with. Maybe a tad more if I was in a good mood at the time.
    I think that's roughly 1.06000 as a percentage. Is that what you meant by a maximum peak of 0.5?

    The file you're scanning with peaks above maximum..... don't forget some lossy formats do store peaks above 0dB without actually clipping them. Are you sure the peaks are physically clipped rather than being clipped on playback? Do they look clipped in Audacity if you lower the volume? Audacity will probably import the audio as 32 bit float so I don't think the peaks would be clipped while it's being imported as such, although I'm not 100% what Audacity does.
    Yeah that's what I mean a maximum peak of 0.5, I remember you saying that lossy can store peaks above 0DB but when i played the movie both on tv and pc the same scenes where distorted really bad, like over volume, I played my normalized audio file as well and the same was happening on tv and pc

    yeah they do look clipped in audacity, there are only about 20 files that need working, will finish them off today then all the movies are done, when i open the audio file in audacity you can tell that they are clipped as you do a scan for clipped sections and then use fix clip, that's what i did, will only use audacity for any i come across in the future as it is time-consuming and with foobar i just set my converts up before bed and let them do there thing.
    Finally, all done now I can move on to my music and just do any movie file when I get them, was a pain as I had so many to work on but now caught up and plex is upto date for my tv, big thanks hello_hello helping me use foobar and getting these video's sorted. much appreciated
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