When I purchase some of my crappy top 40 from Itunes, they are AAC 256 kbps audio files. This makes is hard to analyze for the key in Mixmeister as it won’t recognize them. I am, though, able to convert them to mp3s within Itunes. I have the conversion set to the highest quality…320kbps.
Does it make sense that they can be converted UPWARD from 256 to 320, or are they really still at 256 since that is what I downloaded them at and just say 320?
As I understood it AAC is a better form of compression than mp3, so at 256 is probably equivalent to 320 mp3…
TBH though, I think it’s minimal difference, certainly not one anyone should give much of a hoot over. I doubt many people would be able to notice any difference at all…
it will be pretty close but not a true 320 constant bit rate song. aac 256 is pretty good though so making it into a 320 wont sound to bad. don’t expect a 192 or below file to sound better though.
I disagree. I did a pepsi challenge using a track ripped at 320MP3, 320AAC, 320ATRAC and the original CD played on a really nice system and the difference was astounding. I liken it to Sculpture made out of marble(CD) to the same sculpture made out of Lego’s(MP3). The AAC, I felt, just barely edged out the ATRAC with the original CD being the best. That being said, unless you’re playing different codecs & bit rates together, I would doubt anybody would be able to tell what format you’re playing. To the OP maybe you should re-encode the same track at 320MP3 & 256MP3 and see if you can tell a difference. Just try to listen on a good system and not your computer speakers.
From what I have read, and sorry - no references handy, converting anything from one lossy codec to another is strongly discouraged. I rip in wav and convert to Flac, and buy the same online if I can, admitting to have an aversion to any lossy format.
Never ever convert a song from a lower bitrate to a higher bitrate. When you convert a song to shrink it down it “throws out” information that the human ear can’t hear. So when you convert from a lower bitrate to a higher bitrate your just running it thru an algorithm that throws out some of that information again. Unless you are encoding from a Direct Copy like FLAC or .WAV then you will only do more harm than good.
The previous two posts are spot-on. I’m only repeating the information because it’s really important, and a lot of people don’t understand lossy codecs and transcoding: Never, never take a lossy (MP3, AAC, etc) file and encode it again into another (or the same) lossy format. This goes for increases in bit rate too. Taking a file at 256 and changing it to 320 absolutely does not give you a higher quality file than the original. In fact, you are degrading it. Up, down, keeping the same bit rate, or even going from one format to another it doesn’t matter. Every time a file is encoded into a lossy format, the encoder permanently removes information from the file. Only do it once, if you do it at all.
Side note: Transcoding a lossy file into a lossless one (WAV, AIF, etc) is not damaging your file, you can do this without worry. But keep in mind, that this does NOT improve sound quality either. And once you do it, as before, do NOT re-encode the file into a lossy format.
That doesn’t work either here is an excerpt from a transcoding guide.
Every time you encode with a lossy encoder, the quality will decrease. There’s no way to gain quality back even if you transcode your MP3 128 kbps into a MP3 320kbps. The sound quality of the result will always be worse than the (lossy) source file along with having a larger file size. Even transcoding from a 320kbps CBR MP3 to a 192kbps CBR MP3 will result in worse quality than if you just transcoded directly from a lossless format to 192kbps CBR MP3 in the first place.
You already know what happens when you transcode a lower quality bitrate file to a higher quality file bitrate; the result is a file with a larger file size and no added quality. The same is true of a lossy to lossless transcode. When one downloads a lossless file, he/she expects a bit-by-bit replication of the original source.
Lossless formats include: .WAV, FLAC, ALAC, APE. The first 2 being the most popular. You can transcode any of these formats to any other format and bitrate as they are direct copies of the original.
which makes the platinum notes claims all the more dubious. their algorithms may in cases of a badly screwed file improve things, but for the most part its a pipe dream to be able to apply some magic to a lossy encoded file and suddenly get better quality by decoding, applying magic pixie dust and re-encoding. whatever impovement you just made will probably be lost (and worse) by adding additional artifacts.
i should add that the vorbis orgg guys long ago promised that they would make it possible to reduce the bit rate quality an ogg file without getting worse audio quality than if you would have ripped straight to that lower bit rate. IIRC they called this peeling, but it never came to be. not sure if iTunes recent ability to strip down the bit rate employs technology like that.
Transcodes FAQ
What is a transcode?
Wikipedia says that “Transcoding is the direct digital-to-digital conversion from one (usually lossy) codec to another.” A transcode is any conversion of format.
Which transcodes are bad?
The rules generally allow only a single lossy stage in the encoding process, and it must be the final stage. So FLAC->MP3 would be allowed, but MP3->Ogg and Ogg->FLAC would not be allowed.
What is a lossy encoder?
Most lossy encoders use a low-pass filter when encoding. The filter is set to cut frequencies above a certain point and leave those below. The reason they’re doing it is, that high frequencies are more difficult to encode and hearing is less sensitive in higher frequencies. MP3 encoders at 128kbps will typically use a LPF at 16 kHz. As you raise the bitrate, the frequency threshold raises. At 192kbps the LPF is usually set at 18 kHz or higher. Conversely, lossless encoders do not remove any frequencies from the original file.
Why is lossy transcoding bad?
Whenever you encode a file to a lossy format (such as MP3, M4A (AAC), Ogg, or mpc) information is permanently lost. It doesn’t matter what you do, it’s impossible to get this information back without making a new rip from the original lossless source. If you re-encode it, you are reducing the quality. This applies to any lossy to lossy conversion, so even if you convert from 320kbps to 192kbps, the final file will still sound worse than if you had just ripped to 192kbps in the first place.
It’s also important to remember to verify that lossless rips actually came from an original source. People that download lossless expect it to be identical to the original. There’s no point in people downloading a bigger file just to get another lossy rip.
So how do I verify that my song isn’t a transcode?
The simplest way is to rip and encode it from the original source yourself. That way, you know that there has been only one lossy step (or that the rip is truly lossless, if you decided to do a lossless rip).
You may also use Adobe Audition, Audacity, ect. to look at the spectral frequency display.
What is the difference between FhG and LAME?
FhG and LAME are simply two different MP3 encoders. Both operate in a similar way - using the low pass filter to remove higher frequencies and compress the file. Each encoder uses a mathematical algorithm in order to determine which frequencies to disregard in order to produce the final file, and it is this algorithm that differs between the two (and, in fact, all) MP3 encoders. Most people will tell you that the LAME algorithm is better than the FhG algorithm in that it removes fewer frequencies for the same filesize and produces a “cleaner” encode. At What.CD, we recommend using LAME over FhG. (For more information, read the MP3 Specific Dupe Rules.)
How can I view the spectral analysis of songs using Adobe Audition?
To view the spectral analysis of audio files in Adobe Audition, first ensure you are in Edit Waveform View by pressing the number 8 on your keyboard. Then, go to File > Open and select the file you wish to test. Adobe Audition will open the audio file in the “Waveform View” by default each time, so you’ll need to choose View > Spectral View or press F9 to switch to Spectral View.
On the Linux platform, Audacity can be used to perform spectral analysis. Import the file, select the filename in the new sub-window, and click “Spectrum.” Zooming is accomplished by holding the CTRL key while scrolling the mouse. Note that Audacity also works on Windows or Mac for those people who prefer it.
Common Bitrate Comparisons
The following section contains a list of common bitrates and their audio spectrum. The LAME were all done using dbpoweramp from a FLAC source, and they are all encoded using LAME version 3.97. The FhG were all done using Adobe Audition 1.5 FhG.
Original (FLAC):
128 LAME:
160 LAME:
160 FhG:
192 LAME:
192 FhG:
V2 (preset standard, aka APS in pre 3.94 versions of LAME):
256 LAME:
256 FhG:
320 LAME:
320 FhG:
V0 (preset extreme, aka APX in pre 3.94 versions of LAME):
Typical Webrip: (notice the gap)
Analysis
As you can see, LAME uses ‘full resolution’ up to the frequency threshold, whereas FhG, encodes at ‘full resolution’ up to 16 kHz, and uses ‘low resolution’ at higher frequencies. This is an easy way to tell which encoder was used. At 128kbps, LAME uses a LPF at ~17 kHz and FhG at ~16 kHz. I have included a screenshot of FhG at 128kbps without the LPF. At 160kbps FhG’s LPF is set at 20 kHz. At 192kbps, LAME stops at 19 kHz and FhG encodes upto 22 kHz.
FhG looks like it’s not doing its job right, but if you listen to the 192kbps samples, you can hardly tell which is LAME and which is FhG. At 128kbps, LAME sounds a bit better, more ‘clear’. FhG encoding at 128kbps without the LPF sounds bad, you can certainly listen to the artifacts.
LAME APS will typically use a LPF at 18.5 kHz, whereas APX will go up to 19 kHz.
How to view the spectral analysis of songs using Adobe Audition
To view the spectral analysis of audio files in Adobe Audition, first ensure you are in Edit Waveform View by pressing the number 8 on your keyboard. Then, go to File > Open and select the file you wish to test. Adobe Audition will open the audio file in the “Waveform View” by default each time, so you’ll need to choose View > Spectral View or press F9 to switch to Spectral View.
I’ve seen a lot of discussion here about how to spot transcodes. Many people have suggested using a spectral analysis from programs like Cool Edit / Adobe Audition and looking at the ‘cut-off’ point. There is some disagreement about how effective this is, but those who recommend it suggest looking for cut-offs between 16 kHz as the signature of a 128kbps MP3 source, and 22 kHz (i.e., no cut-off at all) as the signature of a lossless source.
One counter argument to this ‘cut-off’ level method is that the same cut-off which characterises lossy encodes may also be the result of a poor quality recording - a bootleg of a live show or a ‘third world’ vinyl master.
A number of spectral views have been posted and linked to, but nearly all of these have been analyses of entire tracks… which IMHO is NOT the most effective way to use spectral analysis to detect transcodes.
What I haven’t seen anyone discuss is the ‘blocky’ appearance of the spectral analysis of lossy rips which is noticeable only when you zoom in close enough. IMHO this is a more reliable way to detect whether a file which purports to be lossless has in fact been transcoded from a lossy rip, and may even be a useful way to detect re-encodes from lower to higher bitrate MP3s (although this is much harder whatever method you use). However, some consideration of the source material is necessary here, as well. Electronic music, for example, frequently makes use of instruments that use the same technology as lossy audio encoding, so a blocky appearance in the spectrogram might be normal for a lossless source under certain circumstances.
The image below illustrates what I mean. The track (from an album by Philip Glass) was ripped from CD to FLAC and a 1-second sample was saved to 320kbps LAME MP3 and 128kbps FhG MP3 and then in each case saved again to FLAC. The spectral analysis was done at full screen on a monitor with resolution of 1280 x 1024. Each of the three strips below is of the same 0.15 of a second.
FLAC / 320 MP3 / 128 MP3 compared
And here are bigger strips of the three spectral analyses. The zoom level is the same - bigger simply means that what is shown here is around 0.5 of a second - and NOT the whole track!
Well seeing as I only use MP3 or these AAC for my top 40, I guess I will keep them as AAC and stop converting them. That’s what I take from this at least!
Pretty much it’s why beatport charges an arm and a leg for WAV copies of downloads I downloaded a CD normally 12.99 if I wanted WAV’s it came to 72.98 or something close to that.
