23
A WAV file has the potential to hold "more" or "better" data than an mp3. WAVs employ no compression, no loss; they are as close to an exact replica as it is possible to get.
An mp3 employs lossy compression to achieve the smaller data size.
Lossy compression means that information is actually just thrown away if the algorithm decides no-one would be able ...
22
Just four letters: FLAC.
Some explanation / thoughts on the subject
Warning: this includes personal opinions that aren't necessarily mainstream-accepted. See AJ Henderson's answer for a somewhat more moderate view.
I'd first like to say: being pedantic, there is no such thing as a lossless audio file. Audio is an analogue phenomenon, anything digital can ...
10
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 ...
10
FLAC (free, lossless audio codec) is a non-patent encumbered audio codec that utilizes lossless compression to store the audio. There are many other lossless options that support compression, but FLAC is more or less the defacto standard. Since it is lossless, the waveform from it will exactly match an uncompressed wav, however it looks for patterns in the ...
9
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 ...
9
The ultimate answer for this is: It depends on what material you are encoding.
And the strongest scientific evidence is in the coders themselves. When encoding an mp3 using VBR (Variable Bit Rate), some encoders show how many frames were encoded using which bit rate. Here's a screenshot from LAME:
You will notice that only 10 frames out of 10735 were ...
8
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, ...
8
Download Audacity here. Also download the LAME MP3 library here
Install Audacity on your system. Install the LAME MP3 library.
Open Audacity
Click on File > Open... and select the mp3 file in question
Click on Track > Stereo track to Mono
Click on File > Export. Choose "MP3 Files" as Format on the dropdown menu and click save.
Note 1: If you want to save ...
8
No. When you convert a file from .mp3 to .wav, no new information is added: there is no way to regenerate the information that was lost when you created the mp3. All the extra data in the .wav file is redundant.
6
Usually, an application that plays encoded files (be it audio or video) will need to decode that file to a format that the target interface (be it a video or audio interface) can output on a standard port (like analog or spdif for audio, hdmi or vga for video).
Most audio interfaces are waiting for PCM datas. Depending on the OS host of the platform, the ...
5
MP3 Diags is a comprehensive application that can identify up to 50 types of issues with MP3 files and also has tools to fix most of those issues.
5
No. Once lossy formats are encoded, any data not saved within the file is lost. You could convert a lossy MP3 to a WAV or an M4A file but the quality of the WAV or M4A would be exactly the same as the original MP3.
5
MP3 is the 'colloquial' name for "MPEG 1 Layer 3" audio encoding. The purpose of mp3 encoding is to reduce the overall size of an audio data stream whilst maintaining an acceptable level of listening quality.
It is implemented using a "codec", meaning that you need an "Encoding" function and a "Decoding" function in order to listen to the audio. The ...
4
In order to make sound, your computer must drive the speaker with a time-varying voltage. In order to create the time-varying voltage, the computer must send a sequence of numbers to a Digital-to-Analog Converter (DAC).
The simplest .wav file just contains a sequence of numbers that are ready to send to the DAC.
An .mp3 file is a much more sophisticated ...
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.
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
You wouldn't get the same result as you would if encoding from the original source. The losses do compound. But 320 kbps is high enough that a one-generation-removed re-encode to 128 kbps would be perceptually acceptable.
3
What I'm wondering is why does the algorithm decide to put the random values at the front of the first chunk instead of the end of the last chunk? I'm also wondering, why are the random values random? Why can't the encoder just use 0.0?
The noise is almost certainly coming from the decoder not the encoder. The encoder processes the audio and produces a set ...
3
This Winamp blog post describes exactly what you are looking for. The function is called Replay Gain and can be applied as follows (relevant text cited from blog article in case the link goes down):
Select the files you would like adjusted (DO NOT try to calculate
your entire library at once)
Right click and choose Send To -> Calculate Replay Gain ...
3
If you down sample sounds as Stavrosound has said you need to be wary of the Nyquist limit. Roughly the sound must be sampled at twice it's frequency to be accurately represented.
In game sound people will down sample making sure that the majority of the useful data is below the limit, eg. If the audio has no useful info above 4k then the sound can be down ...
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
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.:
http://www.synthtopia.com/content/2010/12/18/free-wavetable-...
3
FLAC being the most popular one, there is a comprehensive list of lossless compression formats on Wikipedia: http://en.wikipedia.org/wiki/Lossless_compression#Audio
Compressed files should be processed by the CPU before being used. This is not preferable in professional editing as CPU is a very valuable and expensive resource than storage space. Since WAV ...
3
This question is a bit out of context on a sound design q&a site, but easy to answer.
Whenever you re-encode an already encoded soundfile (mp3>export to mp3) you lose more 'sound' information. Look up 'lossy' and 'lossless encoding' on wikipedia and you'll understand it better.
Oh and in general it's easier for us to answer if you include more details. ...
3
A sample rate is the rate at which samples are taking from the source sound. It says nothing about how much information is stored in those samples. Whether an audio codec is considered lossy or lossless is dependant on how much information is carried over from the original recorded medium usually (for the sake of argument) based on the 1,412 kbit/s bitrate ...
Only top voted, non community-wiki answers of a minimum length are eligible
