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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 ...
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
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.
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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
Mp3 is audio only. The file extension can be virtually anything; if it's recognized as an extension known by Windows the associated program will be used to open it. The program will usually then read the file header to determine what file it really is, regardless of the extension. If the format can be read by the application it usually opens it just fine.
...
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
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
Opus is a much newer, state of the art technology than MP3. Any time you encode audio into MP3 or Opus, you never get a perfect, exact copy of the original file. There is always some distortion added and some things missing. This is by their design. They are not meant to perfectly preserve all of the information, they are meant to make small files. To answer ...
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
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 ...
3
That depends on what you mean by audible. You can invert the phase of one and add it back to the other and it will play only what the difference between the two is, however you will hear artifacts that might not have been detectable to normal hearing in the original file since they were previously buried underneath other sounds.
It is also important to ...
3
Use the click track plugin in audacity with a tempo of 12 bpm.
This link: Click track plugin
and then export to mp3.
3
This is an anti-aliasing filter. The --preset insane settings must allow aliasing to become apparent.
Your iPhone can't reproduce the frequencies the filter is applied to anyway, and if it could, you wouldn't hear them.
Disabling the filter would likely lead to a very slightly larger file, and some unwanted (lower) frequencies caused by aliasing. An anti-...
3
you could definitely use ffmpeg for this kind of job.
$ ffmpeg -i path/to/your_input.aiff path/to/the_converted_file.mp3
ffmpeg is a huge framework dedicated to multimedia encoding.
Once installed write down the above command line on your terminal,
this should convert your mp3 to the default parameters of ffmpeg for mp3.
For more infos about options you ...
3
If this wikipedia page is to be believed, what you call mp3 and is actually a MPEG-1 audio layer 3 encoded audio file only supports three sample rates :
32 kHz
44.1 kHz
48 kHz
EDIT
As @PkP has mentioned, besides MPEG-1 audio layer 3, there are more recent versions of the encoding, which allow other sample rates :
MPEG2 later (1995) added 24 kHz, 22050 ...
3
There are a number of factors involved here but mainly frequency response, transient response, signal to noise ratio (S/N) and total harmonic distortion (THD).
Frequency response is largely dependent on the size, shape and mass of a speaker cone and there is no such thing as a speaker with perfect frequency response. But most significantly, the size of a ...
3
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 ...
2
Use any decent quality tape deck. Stereo recording tape decks are quite common.
2
Max uses many threads so it can encode an mp3 on single and each core.
http://sbooth.org/Max/
2
You will lose quality due to second generation loss. Even if AAC and MP3 were identical in quality for a given bitrate, the transcode from one to the other would result in a loss because the information each discard is different. It wouldn't surprise me if converting from a 96kbps AAC to even a 320kbps MP3 would be unsatisfactory quality due to the ...
2
This sort of thing changes over time, but currently the most common recommendation you'll find is to simply use iTunes due to its ubiquity - import the file, add the data, drag the updated file into a folder.
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On Linux systems like Ubuntu, you can use ffmpeg:
ffmpeg -i input.mp3 -c:a libmp3lame -q:a 2 -ac 1 output.mp3
where:
-c:a libmp3lame: The audio codec to use
-q:a 2: The audio quality (bitrate), see LAME Bitrate Overview
-ac 1: One audio channel
2
Your understanding of lossy compression is close, but a little bit off. An MP3 (or any lossy compression) doesn't actually reduce the number of samples per second, but rather, it alters the values of those samples.
Lossy and lossless in terms of compression refers to the ability of the compression to reproduce the input to the compression algorithm exactly ...
2
A long-shot, but this might work if you have a similar setup to mine.
It works equally well for DVD [I don't have a BluRay] TV, catchup, YouTube, Netflix, VLC, etc etc
I use my PC's built-in 7.1 analog outs to my 5.1 amp [which saves double decoding & odd phase issues I get over S/PDIF, or HDMI.]
My built-in sound is a Realtek HD device [I know, not ...
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