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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 ...
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.
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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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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 ...
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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.
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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
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. ...
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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
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 ...
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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
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.
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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 ...
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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 ...
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Use any decent quality tape deck. Stereo recording tape decks are quite common.
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I've read lots of blogs where people give advice on adding metadata to WAV files. Most of it doesn't work. iTunes is useless for adding metadata to WAV or AIFF files, because it only sticks to the file inside iTunes. If you send the file to a PC or Smart TV the metadata doesn't stick. Same with VLC player, which I tried as well, it didn't work even ...
2
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
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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 ...
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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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Audible differences is something different than electrically measurable differences. The techniques mentioned (by computing the difference between the two signals) allow to measure differences between two signals but don't give real details about the perceived difference.
The correct way to evaluate audible differences is to setup a panel of listener, and ...
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To my knowledge, this is not possible. To add and crossfade audio data requires to decode the mp3 binary stream to pcm, then re-encode to mp3 the resulting pcm stream if you want an mp3 output.
As mp3 is not very good at cascading, you will (probably) have degradation, the overall degradation will depend on the input and output mp3 bitrate.
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There are programs that can split and concatenate MP3s without re-encoding. So one thing you could do is to split a few seconds off every MP3 at the beginning and end, apply your cross-fades to them (with re-encoding), and then concatenate everything.
Regarding "how much worse" with re-encoding: If you choose a high bitrate for re-encoding, it shouldn't be ...
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