Hot answers tagged mp3
21
Mp3 is "MPEG-1 or MPEG-2 Audio Layer 3 (or III)" and MP4 is "MPEG-4 Part 14". Mp3 is an audio compression and file type, mp4 is a file container capable of holding any video compression (e.g. h.264). So MP4 isn't an extension on MP3 like a bucket isn't an improvement on sand. Read the wikipedia articles for more information.
MP4
MP3
20
If you're after audible differences the only test you can use is a double blind test. Empirical tests such as waveform analysis will most definitely show differences in the output, but that doesn't mean the differences are audible.
In this case you could have some one listen to samples of music and have them guess the bitrate of the samples. The ABX type ...
13
MPEG-1 Layer 3 is complex stuff. I recommend starting your reading here: http://oreilly.com/catalog/mp3/chapter/ch02.html
Another resource that I found helpful to get started with is at: http://www.mp3-tech.org/
In a nutshell, MP3 encoding works by taking a frame (say, 576 samples, which is the smallest frame size for standard MP3) and making a spectral ...
9
There is no way to solve that, and it's because of MP3. This actually isn't the fault of your tools or your technique; this is because MP3 compression necessarily introduces gaps at the beginning of encoded recordings.
It's possible to avoid this problem with compressed audio in general, but you have to use another format:
(from the linked wikipedia ...
8
Although you've already mentioned it, ffmpeg is the canonical (Linux) tool for this. I would recommend invoking ffmpeg directly (from the command line) with something like:
ffmpeg -i [infile].flv -vn -acodec libmp3lame -ab 128k [outfile].mp3
... where
-vn disable video
-acodec libmp3lame convert audio to mp3 using the lame codec
-ab 128k ...
8
What you're thinking of is AAC, not MP4. MP4 is often used as the container for AAC-compressed audio.
AAC uses many of the same principles as MP3, essentially throwing away frequencies that the listener probably won't miss. The main difference between the two is which frequencies each chooses to throw away. AAC was basically "let's take what we learned from ...
6
I find it comes down entirely to the type of music and the speakers you listen to it through:
I can happily listen to pop music on an ipod through earphones at low bitrate (well, I say happily - not really a fan of pop music, but what I mean is it sounds as it should)
If you play the same mp3 through a decent or high end audio system it will sound worse, ...
6
Your equipment is just fine. The differences between a lossless encoding or the original recording and a 320 kbps MP3 encoding are very subtle and most humans cannot distinguish between them. Lowering the bitrate should make the difference more obvious.
Try converting your FLAC file to MP3 (or any other lossy compression) at several different bitrates, ...
6
You'll actually get arguably better quality from your CD.
At best those MP3s will be encoded from the same masters as the CDs, except lossily compressed to MP3, while your CDs are uncompressed PCM audio. Any improvements to the production process - and AFAIK there haven't been any major improvements in quality at the professional level (take that with a ...
5
If you like GarageBand, then consider upgrading to the latest version. GarageBand added the MP3 codec in version 4 (the current version, version 6, is bundled with iLife '11).
Select the "Export Song to Disk..." entry in the Share menu.
In the export dialog, pick "Compress using: MP3 Encoder".
4
Best I've been able to come up with to this one (and I'm interested in a better answer here myself) is a GearSlutz thread on what (if anything) anyone is doing differently to master for streaming audio. The answer from Lupo is relevant and interesting. He suggests:
Level is important. 'Hidden' overshots will be revealed through coding and decoding. ... ...
4
Yes, it is possible, but it isn't easy.
There are a number of free tools for editing audio in the frequency domain. I haven't had much luck isolating specific sounds with them, but I have been able to do some sonic manipulation that wasn't possible with traiditional audio editors.
Tapestrea, from the Princeton Sound Lab.
Spectro-edit
SPEAR
The latest ...
3
Give the Sony M10 a try. Battery life is insane, particularly on a pair of Eneloops.
The manual is available here.
I just ran a test on my M10 - after a power cycle, the M10 remembers which track you were playing, but not the position within the track. The Sony has very good battery life, so you could just slide the power button to "lock" mode and keep it ...
3
I had the same problem. Based on some suggestions in other forums, I tried five easy things. One or more of the changes I made did fix the problem. Here are the things I tried:
Deleted all of the album art from the song files. Do this in a separate folder with copies of the songs, not in your primary music folder on your computer! Otherwise you will ...
3
If reducing your steps is what you're after, have you considered a hardware approach? Something like a Zoom H1 will record a decent stereo image if left in the middle of a rehearsal space and it'll save the data as MP3 right from the outset. All you have to do is download it from the device, upload it to your server. Doesn't really get much more ...
3
You may find a lot of subjective answers here, and some of the quality seems to be dependent on what you play the files on - for example AAC and low quality mp3s will be fine on your iPod through headphones, but pop them on a decent system with good speakers and they will sound crap (Skeptics question here)
Each of the lossy formats has its own way of ...
3
The video may already contain audio in MP3 format. In this case it's better to copy the stream. This avoids re-encoding the audio, which takes time and involves quality loss.
ffmpeg -i [infile].flv -vn -acodec copy [outfile].mp3
... where
-vn drop the video stream
-acodec copy copy audio stream directly to output mp3 file
3
It is highly unlikely there will be any audible difference. The audio quality of an mp3 file is decided by it's bit rate and how good the encoder is.
In almost all cases you want to do as few re-samplings as possible, as it is a bit like making a copy in the analog world. In the case of mp3s, if you have a 96Khz source it has to be re-sampled to either ...
3
When you encode a WAV file to an MP3, some information is irretrievably lost. When you decode the MP3 back to a WAV file, the decoder recreates something close to the original waveform, but not exact due to the lost information. When you re-encode the WAV file back to MP3 once again information is lost. The second MP3 file is of lesser quality than the ...
3
I do (rarely) notice a difference between FLAC and 320 or VBR V0 mp3, especially in high frequencies.
Try this: open a FLAC file with Audacity and then click the track name and click on Spectrogram.
Now convert the same file to mp3 @ 320kpbs and do the same with audacity.
There must be an imaginary line in the mp3 file in >14Khz frequencies, that's like a ...
3
Here is such a synthesizer that allow you to "free hand" draw waves (all synthesizers allow adjusting attack, sustain, decay and so forth so for this purpose you can use about any synthesizer).
Throw this onto a MIDI and VST capable sequencer/DAW such as Cubase, Protools, Reason, FL etc.:
...
3
Take "no" for an answer. (-:
If the voices occupy distinct frequency ranges you might be able to use bandpass filters to separate them. But this is highly unlikely, as most voices have most of their ranges in common.
Switch Audacity to FFT view to see where the voices fall in the audio spectrum.
2
XRECODE does a great job of batch conversion. It supports multiple processors to improve performance. It keeps your tags (but not embedded album covers) during conversion.
The current version is not free, but you can download the previous version for free.
I am currently using the free version on Windows 7.
To convert the files, all you need to do is ...
2
I work with sound for a living.
I have placed the same cut from 128, 192, 256 and 320 into a human ear sound analysis test range with a $100,000 sound sampling oscilloscope and the full wave form across the entire spectrum of the human ear is filled.
Further analysis shows that the main problem with mp3 sound samples in not in the bit but the distorted ...
2
There are some neat studies in the field of psychoacoustics that claim that lossy audio compression tasks our brains more than less lossy formats. I cannot for the love of me remember where I got this but I believe it was from TED.com I did do some background checking on those claims after the fact and they seemed to be fairly accurate.
Most of us can ...
2
To an untrained ear it would be borderline impossible to tell, and even a trained ear needs to pay attention. As such, it is hard to quantify perceptible differences as everyone will perceive it differently, if at all.
My approach to audio engineering, which I picked up from my time at SAE, is to go by feel. You can monitor everything, tweak to the nth ...
2
Try MP3Tag. It is free and allows you to set the tags from the filename.
For your example, you would select all of the songs and use the 'Filename-Tag' command. Enter '%track%-%artist%-%title%' and click OK. This will fill in the track, artist and title tags with the correct information. The album and other tags can be entered manually once per set of ...
2
In my case (Jetta 2007), MP3s encoded in VBR (one of the albums I bought from Amazon MP3) will randomly fail with "Err Title". Very irritating. You can check whether an MP3 is encoded in VBR by right clicking on the track in iTunes and then choose Get Info. The info screen will have Bit Rate field, and VBR will be shown in parentheses if the MP3 is encoded ...
2
No. There is no way to solve this problem on all players with the MP3 format or any other format for that matter. As @Warrior Bob has stated:
MP3 compression necessarily introduces gaps at the beginning of encoded recordings.
Even if you get the silence completely removed it still may have a gap between tracks depending on the player you are using.
...
2
A good place to look for this type of info is Hydrogenaudio. But... because this is so subjective (quality) it ultimately comes down to doing some listening tests yourself.
As a purely pragmatic thing, I'd use .mp3 for lossy compression as it's established itself as the gold standard for better or worse.
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