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 ...
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 ...
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 ...
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) audio gives you:
44100 * 16 * 2 = 1411200 bps or 1411.2 kbps (kilobits per second, i.e. bps / 1000)
To express that in bytes, kilobytes or megabytes use the ...
The most direct, if somewhat "hands-dirty", way of doing this is to manipulate the header with a hex editor. There are two entries in the RIFF header that need to be modified: the sample rate itself, which is 80 BB 00 00 in 32-bit little-endian,
and the adjacent byte rate, 192k or 00 EE 02 00 for a 48k 16-bit stereo file. (96k for mono-16bit, 288k for 24bit-...
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.
Very easy. Just use SoX:
sox -r 44.1k infile.wav outfile.wav
The -r 44.1k overrides the header information in infile.wav and will not perform any effects on the samples themselves.
More info: sox.sf.net
If you want to be technically accurate, a sound file does not have a volume, it has a level. Volume can only exist when there is an actual speaker producing sound as that has a fixed SPL(sound pressure level). A signal can be played back at any volume provided the speaker is able to produce it at that volume.
Instead, it is referred to as signal level ...
Your sine has come out a couple of octaves too low, for some reason. Indeed it's infrasound, 6.6 Hz to be precise, which rather explains why you don't hear anything.
A couple of ways you could have found this out:
Actually the sound is still "audible": over a decent subwoofer, you'll hear (or rather feel) a throbbing similar to a ship engine, and things in ...
WAV files are a realisation of a Microsoft format called RIFF (Resource Interchange File Format), that is specific to audio data. The file contains two Chunks (or blocks). The first chunk a file header that contains format details for the data that is held in the second chunk. The data is "contained" in this chunk and can be encoded using a number of ...
After some more searching, I have found that ffmpeg is able to do this by converting the WAV to raw PCM format, and then converting back to WAV with the correct sample rate in the header.
ffmpeg -i input.wav -f s16le -acodec pcm_s16le output.pcm
to convert input.wav to raw PCM data. Then
ffmpeg -f s16le -ar 44.1k -ac 1 -i output.pcm ...
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 ...
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 expert, but I could try to give you some pointers.
Your first choice is whether to compress or not. The best known example of compressed formats is probably MP3. ...
If you use or have access to a mac, Similarity should be able to help with this. The free version may be sufficient for your needs.
From the site:
Similarity rapidly scans your music collection and shows all duplicate
music files you may have. The comparison powered by "acoustic
fingerprint" technology considers the actual contents of files, not
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 formats do not support 96kHz
dither the 24-bit file to 16-bits because the lossy formats do not support 24-bits, and if you don’t dither, all you are doing is ...
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 ...
One way i do this, using Adobe Audition, is to "Open as" raw PCM data (one has to type an * in the name bar of the opening window to show all files), and then define the sample rate and mono/stereo information in the pop up box that comes up.
This has proved useful for opening files that were stopped while recording in programs like Ableton Live.
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 ...
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 ...
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 can create a .wav or .aiff file from a function that synthesizes a waveform or a complex signal.
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. FLAC seems to have wide support in desktop audio editors, and is a supported playback format on Android devices. The default Samsung Music app should playback ...
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/s/ofzrbiqebut05zk/fixwav?dl=0
Also this is the methodology I used.
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:
At the beginning of a recording, a RF64-aware application will create a standard RIFF/WAVE or BWFwith a ’JUNK’ chunk as the first chunk.
Don't think of it as the "header," since the bext chunk, like any other chunk, can be located at any point in the file. I've seen plenty of .wav files that have the bext chunk after the data chunk.
Getting to the bext chunk simply requires parsing the file chunk after chunk (like you're probably already doing for the fmt and data chunks) until you get to ...
First step when reading audio data from a wav file is to parse the header, especially the 'fmt ' chunk, as described for example in the wikipedia page or on this page.
The 'fmt ' chunk will give you informations on :
the audio format (PCM or not ...)
the number of channels
the sample rate
bits per sample
The audio samples themselves are in the ...
8-bit 8K isn't going to be great quality in any case, but WAV is as good as you can do. Straight uncompressed MS-format (RIFF) WAV is the most compatible format for interchange.
There are two senses of 'compressed' at work here. U (mu)-law and A-law refer to schemes for compressing dynamic range, not for reducing file size. Typically they would be 'expanded'...
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 currently available to measure perceived loudness is using the R128 standard. I hope you're on a Mac, because then you can use this free commandline tool:
To examine difference between files, an hexadecimal editor will be helpful. Most of them allow to display differences between two files. You can find one appropriate to your operating system. A good starting point is to see what are the differences between the two files, especially what is different in the non-playable file.
Now, the point is what to look ...