Here is the deal.
The raw sample (or raw sound) is encoded with a certain quality. Some sound cards can go further with 64-bit sampling.
But suppose we have sound files with a certain KNOWN quality.
CD quality is good for the human ear.
The studio, however, would use better samples. Like a 24-bit standard.
So, you have waveform filename.wav, which really has a sampling frequency of 44100 Hz.
What does it mean?
This means that a computer can receive a huge number of different samples per second to represent almost accurate sound.
Is the sound original? Depends on how it was done. If this was done by your computer and software using a 16-bit standard sound card, yes, it is.
If it were from an analogue recording, then it lost some of its quality when digitized at a frequency of 44100 Hz, fortunately, not so significant for the human ear. NOTE that mp3 recordings is a bad idea for professional recording. But since the mp3 recording does exist ... this adds complexity to your question .: P
Thus, the sound quality is lost when digitizing with a 16-bit sound card. Now this can happen when you encode something in mp3.
Check your photo. Above 17,000 there is no sound. This was done in order to make the sound file significantly smaller without causing significant damage to the sound quality. Is that the same sound? No. It looks like the same thing. But the sound engineer LOVES original and high-quality samples because of information that is NOT cut.
Imagine that I made an original sound so balanced and compressed that even after converting an MP3 it’s hard to say whether it is an original sound or not. Imagine using equalizers to cut any sharp edges and gate effects to normalize it. In addition, my sound generators are some 8-bit oscillators passing through some fx and filters.
If I convert it back to wavetable, there can be no difference.
For instance:
[UNCHANGED FREQUENCIES][CUT FREQUENCIES] Waveform: ================================= mp3: ======================= Waveform: ======================= Waveform: [UNCHANGED FREQUENCIES][CUT FREQUENCIES] Waveform: ================= mp3 ================= Waveform: ================= The following seems impossible to me (except if the converter has bugs thing that can be heard) [UNCHANGED FREQUENCIES][CUT FREQUENCIES] Waveform: ========================= mp3 ======================= Waveform: =============================
So your question depends on the original source that you used in the first signal.
The good news is that the RARELY THAT sample is limited and compressed. Therefore, it seems to me that the CD you are using is likely to sound like the original waveform, while, as you can see, mp3 cut out the frequencies.
Of course, you need a frequency analyzer and spectrum, as MischaNix has already shown.
There are many mp3 encodings. Some of them are static, some dynamic, some reduce more, and some reduce audio information. For this reason, some of them are also larger than others.
Now there are lossless formats. And then there is ogg, which is small enough and also has great quality.
Thus, this issue can become a huge topic for no reason. I will not talk about all these things.
If the problem gives the original sample, your photographs show me significant differences between the two samples. I mean, shaping a waveform from a variation of mp3 slicing should look like this has changed. You cannot get information from nothing.
Burn mp3 to a CD, then get a wave, compare the new waveform with the old and mp3 signal. It will probably not be the same thing, so you can hit the jackpot here. Perhaps you have the original backup at your fingertips.
From now on, try to select the source material and store them on a CD or DVD before discarding them. Or at least keep good uncompressed samples in a backup.
Open questions:
If the spectra were visually indistinguishable, I would not know if there is a real difference or that I simply cannot distinguish them.
Right. But this would rarely happen without sampling intent.
Why ask such a question? :) Do you have steganography? If so, be sure to remember the nature of the sound you are going to use. Samples do not fit. "Ready songs"!
Similarly, what would I do if I didn't have an MP3 file for comparison, just one sample audio?
Since there are many mp3 encoding settings of different qualities, you can check if the lowest quality has been used. If not, there is uncertainty due to compression capabilities. If this applies to the entire sample, you must ensure that compression is required. That is why you cannot be sure of the song. Firstly, you do not record with hard compression SO. I think this is another meta reason why you need natural sound. So if you are lucky about the record. Now about the completed mastered song ... everything becomes rude again. It's about nature, the type of sound. Recording is easier to understand what happens if you knew you were using waveform recording. Of course, recording an mp3 is a waste of time. On the other hand, a finished song, usually these days, burns compressors, limiters, gates and chain compressors. The volume of use of these methods in modern development is huge. So ... you really need luck to find out if the original part was compressed before you started the original waveform.
Is there an automated method that would answer the question with reasonable probability?
No, that I know. Sorry. :( But this does not mean that no one can do this.
BUT!
A stereo sample is usually divided into two channels. Left and right. Now, if you have a spectrum analyzer on a digital workstation, and look only at the left channels of two different samples, you can see on the fly if they are the same or not, I think.
To understand what I mean, see IT . Go at 05:00 and just watch the interface.
Phew Hope this helps you further, as it took some time.: P Greetings.
Edit: fixing some things here and there.