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Convert mp4 Youtube video to lossless audio file
roozhou
post Oct 22 2010, 07:39
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You can also try ffmpeg:

ffmpeg -i input.mp4 -vn -acodec copy output.m4a
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DOS386
post Oct 23 2010, 10:33
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QUOTE (roozhou @ Oct 22 2010, 00:39) *
You can also try ffmpeg:

ffmpeg -i input.mp4 -vn -acodec copy output.m4a


That's what I would try. There should be no quality loss. To test you can "play" the movie and well as the stripped audio-only file with MPLAYER into -vo pcm and the 2 WAV files should be identical (untested).

To get rid of the proprietary AAC codec with no loss, the solution is MPLAYER -vo pcm too and then compress the WAV file with FLAC or WAVPACK. The 100% free and 100% lossless audio-only file might have a size comparable to the original video, or could be even bigger in worst case :shock:


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greynol
post Oct 23 2010, 20:07
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QUOTE (DOS386 @ Oct 23 2010, 02:33) *
To get rid of the proprietary AAC codec with no loss, the solution is MPLAYER -vo pcm too and then compress the WAV file with FLAC or WAVPACK. The 100% free and 100% lossless audio-only file might have a size comparable to the original video, or could be even bigger in worst case :shock:

If you go back and re-read the entire message you will see that this advice is no longer useful.

While you're at it, please tone-down your the open-source zealotry.


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woody_woodward
post Oct 25 2010, 02:18
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QUOTE (320kbp/s @ Jun 17 2010, 01:57) *
Hey guys, I try to convert some of my favorite songs from Youtube video file (.mp4) to lossless music, but the best result is only 320kbp/s.
Could you please show me a software that we can convert mp4 file from Youtube to lossless form like wav or flac, ...
Thank you very much !

I use FLV Extract.
http://moitah.net/#FLVExtract
It will split out the audio and video into seperate files. No transcoding. If the FLV contains MP3 audio, that's what you get. If it contains AAC, that's what you get. You can do your own transcoding, or not.
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Zarggg
post Oct 26 2010, 22:52
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That solution is irrelevant, as the OP is using MPEG-4 containers, not FLV containers.
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Chef
post Oct 31 2010, 16:59
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QUOTE (dv1989 @ Oct 20 2010, 11:08) *
For the trillionth time, there is no such thing as a direct conversion between audio formats. The source file is always decompressed to PCM before being fed into the encoder of the destination file. What seems to confuse some people and lead to questions like yours is that some applications hide this fact.

So, yes, Iím afraid that is a silly supposition. Lossless is lossless; that is, it is exactly the same as the decompressed PCM fed to the destination encoder.

Ah... Sorry. So then why does an mp3 get worse and worse the more times you convert it? If a 192 kb/s file were converted to 192 kb/s, shouldn't it output exactly the same? And yet the report I'm most familiar with, is that it does bug out.

In my head I'm imagining the program has determined what it wants to take out, and will always only take out that information... I've only been able to understand why there is a loss of quality in lossy to lossy conversion by assuming that the file is actually different. Your explanation, while I believe you, presents a mystery ohmy.gif
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devnul
post Oct 31 2010, 17:32
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well thats what I did:
- dl mp4
- use mkvmerge (mmg.exe) to take the audio track
- use mkvextract to get an aac file
- use mp4iptools (mp4creator.exe I think) to create a new mp4 file

it's rather complicated but at least I know what happens to the file, instead of using some freeware converter
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Cynic
post Oct 31 2010, 21:06
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QUOTE (Chef @ Oct 31 2010, 15:59) *
Ah... Sorry. So then why does an mp3 get worse and worse the more times you convert it? If a 192 kb/s file were converted to 192 kb/s, shouldn't it output exactly the same?
No, because MP3 is lossy, not lossless.

QUOTE (Chef @ Oct 31 2010, 15:59) *
I've only been able to understand why there is a loss of quality in lossy to lossy conversion by assuming that the file is actually different.
And it IS different: assume you start with a PCM file that you convert to MP3 once. If you convert that MP3 back to PCM, the result will be different from the original PCM file; even though its size would be the same, the audio data would be different. That's why if you compress that new PCM file to MP3 one more time, you will lose quality again.†
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Zarggg
post Nov 3 2010, 04:37
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QUOTE (Chef @ Oct 31 2010, 11:59) *
In my head I'm imagining the program has determined what it wants to take out, and will always only take out that information... I've only been able to understand why there is a loss of quality in lossy to lossy conversion by assuming that the file is actually different. Your explanation, while I believe you, presents a mystery ohmy.gif


Here's a dumbed-down explanation (it's not EXACTLY right, but close enough for people not concerned with the details):

MP3 encoding uses an algorithm known as the "psycho-acoustic model" to analyze an audio file and determine what pieces it can take out while still producing a file that sounds the same (or "close enough" to the average ear). If this algorithm is applied using insufficient bits for the task (i.e., low-bitrate files), or applied several times in succession to its output (i.e., re-encoding a lossily-encoded file), then it can "mess up" and generate what are known as "artifacts": warbling, "underwater-like" sounds, blips, pops, etc.

In other words, it doesn't "know" what bits it's going to take out before-hand; it determines that based on the input. It won't (and can't) produce identical output with different inputs.

This post has been edited by Zarggg: Nov 3 2010, 04:40
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psix
post Dec 30 2013, 21:45
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QUOTE (mixminus1 @ Oct 20 2010, 22:09) *
FWIW, personally, I've only processed a handful of files using the "-single" method (about a dozen), and thus far, I haven't encountered any problems outside of iTunes not being able to read the bitrate.

iTunes can play them just fine, and foobar2000 also plays them fine and displays the correct bitrate.

I'm a little late to the game, but Rockbox, for example, doesn't like all -single files and I think I found out why:
-single seems to retain the trackid, which causes problems when it's not 1.

Then again I do have a couple of files with trackid other than 1 with odsm and sdsm tracks present in root OD, which do work.
They were created with mp4ui, or that's what mp4box tells me - I can't remember.
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