What are the advantages of WAV vs. MP3?
Wave is an uncompressed or lossless format, while MP3 is compressed or lossy. Technically .wav is just a container format and can contain various types of compressed or uncompressed audio, but normally you will see that it contains uncompressed LPCM audio (same as on audio CDs). With .wav files, you essentially get a raw bitstream representation of the audio signal in digital form. An analog sound produced in the real world essentially contains an infinite amount of information because it is a constantly changing wave (see below). To bring these sounds into the digital domain, you need to sample the signal at various intervals to approximate the sound. For .wav, the audio signal is generally sampled at 44,100 times per second or more, and each sampled value is recorded so that the sound wave can be played:
MP3 files are compressed to compress the same audio information into a smaller file size. The .wav format is ideal for very faithful representations of the analog signal, but as you probably know, that usually costs larger files. Compressed audio (and video in a similar way) is designed to reduce file size while maintaining a respectable level of fidelity. In simple terms, compression tries to remove unnecessary data from the stream and reduce the signal to its most necessary components. With MP3, compression and encoding algorithms use a model of how we listen to analyze audio in the frequency domain and remove any unnecessary information. For example, due to auditory masking if there are two sounds at close frequencies, we will often only hear the loudest if the volume difference between the two is significant. So for MP3, the lower volume sound could be ruled out and the audio would sound essentially the same to our ears. Learn more about the technical side of MP3 encoding here.
In practice, both .wav and MP3 have their uses. For production, .wav is the standard because it will almost always be a 100% accurate, bit-by-bit reproduction of the source material. MP3s can be a decent alternative at high enough bit rates. Bitrate is a measure of how many bits per second MP3 encoding will use, which means that the higher the bitrate, the closer the MP3 will be to the original uncompressed stream. Bit rate is generally measured in kilobits per second (kbps). I like the high audio quality for my digital music collection, so when I have a choice, I generally encode MP3 at constant 256 or 320 kbps. That’s the upper end of what MP3s are capable of, and unfortunately a lot of digital music isn’t encoded that high. When the bit rate drops, it can generally be heard first at the high frequencies, for example, the cymbals of a drum kit will sound. 160kbps is tolerable, but somewhat lower than that and you will really start to notice it. But then again, with a high enough bitrate, the differences between MP3 and .wav are barely distinguishable, especially for an untrained listener (most listeners).
For .wav files, we mainly look at the bit depth and the sample rate. Bit depth is the number of bits used to encode each sampled value. The sampling rate indicates how many times per second the audio is sampled. CD (.wav) and MP3 are encoded at a sampling rate of 44,100 Hz (Hertz means “cycles per second”). Newer computers and audio hardware / software are now accommodating higher sample rates, including 48kHz or 96kHz. For .wav, the bit depth is usually 16 bit or 24 bit on newer systems. For most purposes, when using .wav, 16-bit, and 44.1kHz is sufficient, but if you have the capabilities, it’s generally worth upgrading to 24-bit, 48kHz.
Some sample file sizes for a five-minute stereo recording:
.wav, 16 bit, 44.1kHz: 50 MB
.wav, 24 bit, 48 kHz: 82 MB
.wav, 24 bit, 96 kHz: 164 MB
MP3, 128 kbps, 44.1 kHz: 4.5 MB
MP3, 192 kbps, 44.1 kHz: 7 MB
MP3, 320 kbps, 44.1 kHz: 11 MB
FLAC, 24-bit, 44.1 kHz: 28 MB
FLAC, 24 bit, 48 kHz: 31 MB
FLAC, 24 bit, 96 kHz: 61 MB
There is also a variable bit rate option for MP3 encoding, which should offer slightly smaller file sizes for the same quality. It uses a coding scheme that changes (varies) the bit rate for different parts of the song depending on the complexity and how many samples would be needed to faithfully recreate a given section.
