True FLAC vs. Fake FLAC Options

[XeR0]

I know the topic title sounds absolutely absurd but hear me out. I've tested my FLAC collection by encoding them into V0 MP3. I then decoded the MP3 back to WAV and then compressed it in FLAC. My question is: unless you've ripped the files yourself, how would you know the FLAC file you have is ACTUALLY lossless instead of an MP3 converted into FLAC? I've used the TEST option in FLAC frontend and it doesn't give a result. I have used Audiotester and it does say the file failed because it's TRUNCATED.

Bottom-line: Is there a sure-fire way of knowing that a FLAC file is truly lossless and not a derivative of a lossy file?

This post has been edited by XeR0: Jun 29 2011, 19:59

[Joseph93]

3 things:

1. hi I am new

2. I recently stumbled across a paper which details an algorithm that, with a very high success rate, guess the bit rate of an audio file just using data from the file's high-frequency spectrum. If developed further it could remove the need to visually inspect a spectrogram etc. and would be much faster.

http://www.fileden.com/files/2009/2/14/232...20Frequency.pdf

3. If anyone has Max, I threw together a patch the other day which allows you to compare the quality of two audio files in realtime (by looking at their spectrograms; they play simultaneously). I successfully used it to tell the difference between files of varying bitrate, even between 256kbps and 320kbps mp3s. you can also use vst spectrograms with it if you prefer. If anyone is interested I can upload it

This post has been edited by Joseph93: Oct 12 2011, 05:14

[saratoga]

3 things:

1. hi I am new

2. I recently stumbled across a paper which details an algorithm that, with a very high success rate, guess the bit rate of an audio file just using data from the file's high-frequency spectrum. If developed further it could remove the need to visually inspect a spectrogram etc. and would be much faster.

http://www.fileden.com/files/2009/2/14/232...20Frequency.pdf

In order to obtain the feature data, the source MP3 files were each
decompressed into a 1411 kbps WAV file using the Fraunhofer
IIS MP3 Surround Commandline Decoder V1.4 [2]. This was
done because audio files in this format can easily be read into
MATLAB, and as we have demonstrated, transcoding to a higher
bit rate does not affect the frequency characteristics of the audio
which we are observing.

It is, in my opinion, not a good sign when the author of a paper does not understand that WAV is a lossless format and so resorts to arguing that "transcoding" to PCM probably doesn't change the audio. Regardless, all that paper demonstrates is that if you know that LAME 3.97 was used with default lowpass for each bitrate, you can figure out the source bitrate by looking at the lowpass setting.

[Joseph93]

It is, in my opinion, not a good sign when the author of a paper does not understand that WAV is a lossless format and so resorts to arguing that "transcoding" to PCM probably doesn't change the audio. Regardless, all that paper demonstrates is that if you know that LAME 3.97 was used with default lowpass for each bitrate, you can figure out the source bitrate by looking at the lowpass setting.

So does the paper actually not do what it claims it does? I don't see any reliance on prior knowledge concerning the "history" of the file in question.

[saratoga]

So does the paper actually not do what it claims it does?

It does what they claim, take a known encoder and version and then determine what bitrate was used. It doesn't do what people in this thread are interested in though.

I don't see any reliance on prior knowledge concerning the "history" of the file in question.

Suggest reading section 2, "procedure". They train their model using the same encoder and settings they will then attempt to detect. Without this the system is useless.