26
votes
Accepted
Does the .wav file created from a .mp3 file have better quality than the .mp3 file itself?
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 ...
- 8,560
11
votes
Accepted
How do you compute the bitrate of a WAV file?
Here is how you calculate the constant bitrate (CBR) of uncompressed audio:
Bits Per Second (bps) = Sample Rate (Hz) * Word Length (bits) * Channel Count
Which for 44.1 KHz, 16 bit, 2 Ch. (stereo) ...
- 3,782
8
votes
Does the .wav file created from a .mp3 file have better quality than the .mp3 file itself?
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 ...
- 1,578
6
votes
Accepted
What is the best WAV encoding and why?
I assume that you export sound in order to import it somewhere else. This process has a lot of details and used to be done in the step called "mastering" in the old recording days. I'm no ...
- 1,539
5
votes
Does the .wav file created from a .mp3 file have better quality than the .mp3 file itself?
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 ...
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4
votes
Is there any tradeoff to get the best quality/file size ratio for mp3 conversion?
To get the best audio quality in your final lossy-eccoded audio file output at the smallest possible file size, you have to do 3 steps:
sample rate convert from 96kHz to 44.1kHz, because the lossy ...
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4
votes
Does the .wav file created from a .mp3 file have better quality than the .mp3 file itself?
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-...
- 141
3
votes
Accepted
Tool for creating a WAV given a function of time/sample
The first thing that comes to mind is a tool like Matlab (from the Matthworks). You can download a free trial, but you would have to learn some prograsmming (or Matlabs language). But I believe it ...
- 101
3
votes
alternative to mp3
During the editing & collaboration phase, it's good to avoid lossy formats if you can.
You might want to try FLAC, a lossless format that compresses audio files to about half the size of WAV. ...
- 31
3
votes
Accepted
How can these wav/mpeg files be the exact same duration—down to the (apparent) millionth of a second?
Digital audio is sampled i.e. made up of discrete chunks, with each chunk representing a small sliver of time. So the duration will be quantized to multiples of 1/sampling frequency i.e. if sampling ...
- 694
3
votes
Can I recover an incomplete .wav file?
The problem you have is that the WAV header has not been created correctly. If you are working on a Mac then you can try this application which I wrote to fix broken WAV files. https://www.dropbox.com/...
- 8,835
3
votes
How can WAV files cause any clipping red in DAW meters if floating point can represent much higher values than 24 bit?
Building on Tetsujin and ZaellixA responses:
If your wav isn't itself clipped, its encoding (if samples are floating points or are integer numbers, it's bit depth...) may be missing or wrong in the ...
3
votes
How can WAV files cause any clipping red in DAW meters if floating point can represent much higher values than 24 bit?
0dBFS in floating format is a bit of a misnomer. It's not "full scale" in floating point, instead it's actually defined as the value 1. This is the value that when converted to fixed point ...
- 132
3
votes
GUID difference in R64 and WAV files
The two formats seem to be similar, and as the specification says the older wav-format is compatible with RF64.
From what I can see on p 10, the GUID isn't the only thing that identifies the format:
...
- 466
2
votes
How to ensure uniform sound levels in multiple WAV files?
You are talking about perceived loudness. This is quite unlike peak level. What you can do is measure the perceived loudness for all your songs and then adjust the gain accordingly. The best way ...
- 104
2
votes
Accepted
Normalizing many small WAV files?
I use Sound Forge for this kind of thing. It has a feature that lets you specify a list of processes to run your sounds through, parameters to use, file formats and locations to save to, etc. It will ...
- 146
2
votes
Accepted
Artifacts when exporting sound as 8-bit WAV
Given that your input file is 44.1 kHz/32 bits/stereo file and your desired output is 22.050 kHz/8 bits/mono file, there are actually three processes that are involved in your request :
resampling
...
- 2,834
2
votes
alternative to mp3
While lossless compression like FLAC would solve your storage space issue, it is a heavy load on your CPU, so you may run into problems if your machine is at at all old.
The usual recommended ...
- 5,973
2
votes
Accepted
Methods for decreasing exported .wav file size
WAV is a lossless raw format, so the short answer to your question is no.
The only way to reduce the filesize is to :
use a different export format, or
shorten the length of the project.
There are ...
- 8,835
2
votes
Does the .wav file created from a .mp3 file have better quality than the .mp3 file itself?
You cannot get detail from nothing
Even though WAV files can hold more detailed sound, that doesn't mean they always do. Converting from MP3 to WAV would be one such scenario, the converter cannot ...
- 121
2
votes
How can WAV files cause any clipping red in DAW meters if floating point can represent much higher values than 24 bit?
As Tetsujin states in their comment, any decently developed/implemented DAW will use floating-point arithmetics to increase their internal calculations headroom.
This does not mean, in any case, that ...
- 339
2
votes
How can WAV files cause any clipping red in DAW meters if floating point can represent much higher values than 24 bit?
Usually, 0dBFS is the maximum level which doesn’t clip on an audio interface. A DAW will typically map it under the maximum float level, then if you increase its level in a filter then reduce it ...
- 627
2
votes
from Flac to wav
FLAC is lossless compression. During compression, you can set a compression option: setting this to a higher level creates a smaller file that is still lossless by taking more time to process the file....
- 1,578
1
vote
How do I generate a 0 dBFS sine-wave in SoX? (output clipped warning)
Check out the hexdump of the file you created.
When you do, you will realise that there are 7FFF values in the file that correspond to your sample peaks.
But, I'm going to say that you also have ...
- 8,835
1
vote
WAV file not in sync with source MP3 file
Try using Logic, Pro Tools, or Ableton Live to convert to WAV. The time code should remain synced. Let me know otherwise.
I’m guessing that maybe some setting is being changed, like kHz or bit rate....
- 136
1
vote
alternative to mp3
You can try OGG Opus, which seems to be getting more popular recently. I haven't used it myself, but from what I've heard, it's a little bit better than MP3. Here it is compared with other formats in ...
- 11
1
vote
Tool for creating a WAV given a function of time/sample
If simple waveforms are enough, Audacity is a (free and) convenient tool to go for. In the version I use (2.1.2), the following are creatable per default (please see below for plugin add-on):
Simple ...
- 281
1
vote
Add .wav metadata in Kid3 on Ubuntu
Although it seems that it doesn't solve the OP issue, there's a thread on Kid3 discussion forum that is titled :
Tagging of WAV issue - Music players are unable to read the tags of WAV
Here's an ...
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