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26

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 ...


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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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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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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 ...


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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 ...


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I recommend you to start with pulse-code modulation files (.WAV, .AIFF, etc.): https://en.wikipedia.org/wiki/Pulse-code_modulation What is stored in a typical CD-quality PCM file is nothing more than a few metadata markers, and a trunk of stereo data kept as numbers(note, they may be floating numbers or integers). You can view PCM files as a "dump" from ...


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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 just make up the extra added details to add to the WAV files. It can only remove existing details when compressing. Convert a decent audio file to 16kbps MP3 and ...


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An external ADC/DAC chain might sound better than if you did it in the PC. Is it really? Possibly, or it could be apophenia or other psychological improvement. You would need to do controlled testing with many people to be able to say what actually happened quality wise.


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Supported codecs and containers for the audioread function of matlab can be found at: https://www.mathworks.com/help/matlab/ref/audioread.html


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These spikes are high frequency components contained in the resulting signal decode. There are two possible reasons for this. Method one uses a low pass filter which removes these components. Method two has a decoding fault. Both of these suggestions are speculative as there is no reference signal to compare.


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No. This is about data resolution. Converting WAV to MP3 gets your data compressed, with loss. Converting MP3 to WAV gets your data expanded, without loss (But the loss presumably already has taken place beforehand, so you gain no additional resolution.


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Nope. Data (waveform) and Encoding (.mp3 / .wav) are things independent of each other. The same waveform encoded in both mp3 and wav, will produce a different decoded waveform when run thru the proper decoders. As far as format conversion goes... Wav data encoded to mp3 will lose waveform details Mp3 data encoded to wav will not lose waveform details


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