REVOLVER 2022- WHAT'S NEW?

[Tex]

The spectrum method just discerns whether there has been MP3 compression which produces visible rectangular or square HOLES according to what I looked at. Having nothing above a certain frequency does not indicate MP3 compression it just means the source has limited quality/frequency range.

I listen to 320 bit MP3s in the car but at home I listen to mostly WAVs.

[Lord Reith]

In Audacity, would that be Analyze>Plot Spectrum? And if we see purple all the way from from 50 to 20000 I guess that's a good thing? I notice most mp3s fall short of the 20000 to varying degrees but so do some FLACs I checked, maybe depending on the original type of audio content?

As for mp3/320, I agree they sound quite good in general. I mash all interviews and documentaries that way using XLD, also cover versions and some artists' live shows.

On the left of a track in Audacity you can select the display to either "waveform" or "spectrogram". The latter is the setting you use to analyse an audio in detail. It shows the spectrum vertically in colours and you can zoom in on different frequencies by left clicking the magnifying glass on, say, "2KHz" on the list of frequencies on the left. Right clicking zooms out.

You can also remove different bits of the spectrum using the "spectral edit multi tool" in effects.

It's pretty difficult to spot an mp3 unless it's low bitrate. In that case there will either be a sudden cutoff at maybe 18khz or a blocky look to the higher frequencies or holes in the upper frequencies.

If the recording is in mono but the file in stereo, then you can OOPS the file, amplify it as much as it will go and listen for the swishing, twittery artefacts that characterise a low bitrate encoding. Incidentally, folding down a mono recording presented as a stereo mp3 will cancel out these artefacts. So if you have a mono track like a BBC track presnted as a stereo mp3, fold it down to a single channel for better quality.

The main reason not to demix from lossy source is that the encoding confuses the demixing algorithms. Noise reduction also confuses them. For best results you want a completely unprocessed wave file.

[alphabeatles]

In Audacity, would that be Analyze>Plot Spectrum? And if we see purple all the way from from 50 to 20000 I guess that's a good thing? I notice most mp3s fall short of the 20000 to varying degrees but so do some FLACs I checked, maybe depending on the original type of audio content?

As for mp3/320, I agree they sound quite good in general. I mash all interviews and documentaries that way using XLD, also cover versions and some artists' live shows.

On the left of a track in Audacity you can select the display to either "waveform" or "spectrogram". The latter is the setting you use to analyse an audio in detail. It shows the spectrum vertically in colours and you can zoom in on different frequencies by left clicking the magnifying glass on, say, "2KHz" on the list of frequencies on the left. Right clicking zooms out.

You can also remove different bits of the spectrum using the "spectral edit multi tool" in effects.

It's pretty difficult to spot an mp3 unless it's low bitrate. In that case there will either be a sudden cutoff at maybe 18khz or a blocky look to the higher frequencies or holes in the upper frequencies.

If the recording is in mono but the file in stereo, then you can OOPS the file, amplify it as much as it will go and listen for the swishing, twittery artefacts that characterise a low bitrate encoding. Incidentally, folding down a mono recording presented as a stereo mp3 will cancel out these artefacts. So if you have a mono track like a BBC track presnted as a stereo mp3, fold it down to a single channel for better quality.

The main reason not to demix from lossy source is that the encoding confuses the demixing algorithms. Noise reduction also confuses them. For best results you want a completely unprocessed wave file.

Thanks for your explanation!

I found the spectrogram option, got that going, But the magnifying bit while clicking on frequencies doesn't seem to do anything (which magnifying glass? Any of them just zoom the view).

Also, this scale goes 0-8k or 1-7000 (etc.) depending on setting (?).

mp3s and FLACs both fill the spectrogram up to the top, so the answer about lossy vs. lossless is not obvious to me.

[unwantedman]

and the spoken intro by Ringo on Yellow Submarine is still missing 15 seconds, just like the version on the Real Love CD single released in 1996. Maybe the 15 seconds extra wouldn't have fit on the CD?

where can you find this version with the extra 15 seconds? it's not on the PC Deluxe Edition, and that seems the most logical place if it circulates.

[Tex]

I found the spectrogram option, got that going, But the magnifying bit while clicking on frequencies doesn't seem to do anything (which magnifying glass? Any of them just zoom the view).

Also, this scale goes 0-8k or 1-7000 (etc.) depending on setting (?).

mp3s and FLACs both fill the spectrogram up to the top, so the answer about lossy vs. lossless is not obvious to me.

Set the spectorgram to stop at 20khz which is the limit on CD WAV files. You would want to examine the range like 15khz to 20khz

The format is irrelevant any audio format can have frequencies up to 20khz. MP3 compression is about saving bits not cutting frequencies.

The MP3 codec selectively manipulates the waveform above 15khz or 16khz mainly in the busy parts of the songs as I recall. The codec basically rewrites the waveform to remove details that are not very perceptible but still use a lot of bitrate. There's a lot more data in a WAV file than is actually necessary to produce sound.

The FLAC compression "format" simply encodes (like ZIP) receptive data as smaller pieces to shrink the file size. An analogy is you can take the most common words and letter combos in text file and replace them with a single special characters. Then when decompressing the special characters are swapped out the original things they represent. That is LOSS-less compression because you always get back the original file.

The MP3 codec is actually removing details to get a smaller file size. The trick to the codec is removing and manipulating in such a way that the casual listener can't tell the difference. There's a huge amount of audio perception research behind the MP3 codec and others.

[alphabeatles]

and the spoken intro by Ringo on Yellow Submarine is still missing 15 seconds, just like the version on the Real Love CD single released in 1996. Maybe the 15 seconds extra wouldn't have fit on the CD?

where can you find this version with the extra 15 seconds? it's not on the PC Deluxe Edition, and that seems the most logical place if it circulates.

These are two copies that seem to be from the Real Love single with Ringo's 15-second intro (maybe not the one you're looking for?). The VooDoo version has an extra repeat of the chorus at the end for some reason.

hxxps://drive.google.com/file/d/1W8h0V5mjoBIQ6YWv0fyTslBkAGBXaaa9/view?usp=sharing

[Rupert Pupkin]

In comparison to Lord Reith "real live tempo" of "Rain" (I think it's on his suite Evolver) - he posted this recently - is not that fast. I don't think that Ringo could not play that fast. But it's a freaking faster tempo. I was just wondering how "accurate" this "real live" tempo is, because the official stuff is not something you can 100% trust. Perhaps later someone like csnyfan can give us his expertise, and some mix.

Same thing crossed my mind. It sounds too fast.

I wonder if anyone could analyse it and determine what key it was actually played in. Might be tricky to do - guitars can have capos and drum and bass not easy to judge.

Alternatvely, we just trust it was copied from original studio tapes recorded at that tempo.

I would never have expected them to play in the key of Bflat, because of the way the guitar is played John and George must be using capos (that's a band that goes on the neck to make the notes higher, like George uses on If I Needed Someone in Tokyo). When I was mixing that fast version I just assumed it would be in the key of A because they are playing open chords and at that pitch it is already breakneck speed. But who knows.

I don't think that they could not play it that fast; but did they have to play it that fast to get the strange "slow down" effect they would get when slowing down the tape so that John could record his vocals ?
This required, as you wrote to use capo for guitars, but bass too ? because it really sounds somehow "unnatural" to me (like it is speed-up) because the tone of Paul's bass is really high.
What if ? I'm not sure of that but I don't totally trust the official release (they messed up in the past) what if they adjust the pitch in order to reconstruct the "real live" tempo of the original speed just based on the count-in voice by Paul at the beginning ??
What if for one reason or another Paul's voice and count-in does not belong to this take ? (they probably had to play it several times to get a perfect take; so that would imply multiple false-starts and attempt; thus a lot of count-in by Paul)

regarding the fuzz guitar for Got To Get You In My Life which sounds great and dry I was not thinking of Satisfaction for the Rolling Stones. The Stones could have been influenced by "Rain" and the Revolver album tone for some songs like "Child Of The Moon".

There is a great pdf booklet with the digital release. 51 pages with some photos I think are new. There is a track by track presentation of each song. I have to read carefully what they say about the recording of "Rain" and I'm curious to read "Yellow Submarine".

[alphabeatles]

I found the spectrogram option, got that going, But the magnifying bit while clicking on frequencies doesn't seem to do anything (which magnifying glass? Any of them just zoom the view).

Also, this scale goes 0-8k or 1-7000 (etc.) depending on setting (?).

mp3s and FLACs both fill the spectrogram up to the top, so the answer about lossy vs. lossless is not obvious to me.

Set the spectorgram to stop at 20khz which is the limit on CD WAV files. You would want to examine the range like 15khz to 20khz

The format is irrelevant any audio format can have frequencies up to 20khz. MP3 compression is about saving bits not cutting frequencies.

The MP3 codec selectively manipulates the waveform above 15khz or 16khz mainly in the busy parts of the songs as I recall. The codec basically rewrites the waveform to remove details that are not very perceptible but still use a lot of bitrate. There's a lot more data in a WAV file than is actually necessary to produce sound.

The FLAC compression "format" simply encodes (like ZIP) receptive data as smaller pieces to shrink the file size. An analogy is you can take the most common words and letter combos in text file and replace them with a single special characters. Then when decompressing the special characters are swapped out the original things they represent. That is LOSS-less compression.

The MP3 codec is actually removing details to get a smaller file size. The trick to the codec is removing and manipulating in such a way that the casual listener can't tell the difference. There's a huge amount of audio perception research behind the MP3 codec and others.

Thanks again!

If one of you could post a screenshot showing lossy vs. lossless for the same song to show what to look for, that would be helpful.

[Tex]

Is the other version of "Got To Get You Into My Life" essentially an instrumental mix of the final version or does it have different stuff?

[Ants]

Same thing crossed my mind. It sounds too fast.

I wonder if anyone could analyse it and determine what key it was actually played in. Might be tricky to do - guitars can have capos and drum and bass not easy to judge.

Alternatvely, we just trust it was copied from original studio tapes recorded at that tempo.

I would never have expected them to play in the key of Bflat, because of the way the guitar is played John and George must be using capos (that's a band that goes on the neck to make the notes higher, like George uses on If I Needed Someone in Tokyo). When I was mixing that fast version I just assumed it would be in the key of A because they are playing open chords and at that pitch it is already breakneck speed. But who knows.

It sounds like…in my opinion…that Rain(and other tracks) were recorded with tape machine running at a slower speed…and when you play it back at the proper speed it sounds like they were using a white substance while recording…it sounds ridiculously fast and did they really play it that way?? I’m not of the belief that they did. I think the book documents the use of different speeds while recording

Just listen to the count-in: it sounds natural that way, so the speed must be right. Yes, it's really fast.