The frequency range that the human ear can perceive ranges between 20Hz to 20kHz. Using WAV files to record your podcast will capture this entire range. When MP3 files are compressed, you may lose some sound on the lower or higher ends of the frequency spectrum, as MP3 files cut off around 18KHz. Because of advancements in recording technology, you can easily use a home studio interface and WAV audio to produce high-quality recordings. In fact, many popular home studio audio interfaces on the market provide recording rates up to kHz!
If you're recording your podcast at home, this is something that you should consider. The high quality of WAV files comes at a cost—larger file sizes. These large files can affect your podcast budget and limit how listeners access your show. You might need to spend more money on your show to host and store larger WAV files. That means budgeting more for your podcast hosting app. Plus, you'll have to pay closer attention to the size limits your hosting company sets on file uploads and downloads.
Larger WAV files make them impractical to use with some streaming services and portable devices. This can limit how your audience can listen to and download your show on iTunes, social media, and other subscription platforms. Some listeners prefer to download podcasts and listen to them on their devices at their leisure when offline. Larger file sizes can make downloading and storing your show a hassle, which could turn off listeners and make them tune out.
Consider the content of your show, your podcast budget, and your audience. While there is no cut and dry answer when comparing WAV vs MP3, if you understand the unique needs of your show, you can make a well-informed decision.
If your content mostly involves people having conversations with some sound effects, MP3 will work just fine. WAV formats really make sense for music production. Because of their larger size, using WAV files for your podcast can get pricey when it comes to hosting.
You may want to invest in WAV if you consider audio quality a top priority, and you have the budget. You can't have a successful podcast without an audience. Pay attention to your branding and how your listeners consume your show. MP3 files are smaller, which makes them easier to download or stream, and guests can have greater access to your podcast across all of their devices and websites.
The size of WAV files can put limitations on how your show is distributed, which in turn affects how many people you can reach. Compressed MP3 audio files let you store and distribute your show more easily.
But, you sacrifice audio quality lost in the compression algorithms. You can edit uncompressed WAV audio files more easily. They provide a much higher audio quality, but their size makes them difficult to store and stream. Keep in mind that uncompressed audio files can always be compressed. You can record and edit the uncompressed WAV format, and then compress the file into MP3 format for easier distribution.
If you do a lot of editing after you record audio, and you don't mind the loss in audio quality, you may want to look into downloading a program that converts WAV files to MP3 files. Like with AAC files, using WMA to record and distribute your podcast could cause hang-ups due to device and platform compatibility issues.
This file type is compressed but is still a lossless file type, as little quality is lost due to audio track compression. This file type gives you similar sound quality to a WAV file while taking up half the storage space. Storage is so affordable these days, there is no longer an argument for saving a lower quality file.
Fact 3. An MP3 is lossy and compressed. A WAV is lossless and uncompressed. Fact 4. MP3 and other lossy formats exploit general human hearing to reduce file size. That was the only reason for it to be used, thus causing quality loss. Perceptable hearing depends on the user and the amount of compression used. Fact 5. Fact 6. Fact 7.
It is still the preferred format for all audio included in apps that are released on the Mac and iOS App Stores, as well as Nintendo and PlayStation products.
This results in smaller file sizes, all while delivering higher audio quality. When deciding which audio format is right for you, the first question you have to ask is whether the file needs to provide uncompressed audio or can it be in a compressed audio format? Additionally, following the Red Book standard to provide for the ultimate listening experience, CD-quality audio should use uncompressed audio files at On the other hand, if your intent is to make sharing your music easy and fast, choose a compressed audio format that will provide you with small file size.
At that point, convenience will always win. Additionally, since email providers limit attachment sizes and smartphones have limited storage space, any audio format that can offer smaller file sizes is going to be a winner. For all of those instances, choose a compressed audio format like MP3 or M4A. The smaller the file, the worse the audio quality will be. Well, that will really depend on your use case.
For starters, the historical prevalence still stands today. The great news is, regardless of which of the two formats you choose, you will achieve exactly the same superb audio quality.
The majority of desktop and mobile devices sold nowadays come with native support for MP3 and M4A files alike. For higher quality results, I recommend you choose M4A, which can offer higher sonic results at the same settings, all while still resulting in smaller file sizes than MP3. On the other hand, if guaranteed compatibility is what you need most, MP3 will probably be the wiser choice of the two.
I hope that this guide was able to shine some light on the difference between the four basic audio formats and when to use them. Most modern DAWs allow you to bounce your song in multiple formats at once. Additionally, if you have your song bounced in at least one Uncompressed Lossless format, there are plenty of great audio converters on the market that will allow you to convert your song into any of the other audio formats when you need them. The audio world is filled with many options, and the four basic formats above are just a few of over a dozen different audio formats.
Learn the basics of digital audio and how a computer handles sound, from audio sample rate to bit depth. Often confused with each other, phase-based processors offer interesting sound design options for producers.
Today, we discuss chorus, flangers, and phasers. Often overlooked, convolution is a powerful process for both standard processing and sound design. An analog to digital converter takes thousands of snapshots per second to capture the full audible frequency range of 20 Hz to 20 kHz. WAV files are lossless and uncompressed which means they lose no quality from the original recording. A stereo, CD-quality recording Increasing to 48 KHz and 24 bit stereo will be reflected in a change from 10 Mb per minute to We can now take a closer look at some of the differences between the two.
As I touched on in the overview, lossy compression discards some data from the original recording. The algorithm makes assumptions on what to discard based on frequencies the human ear is unlikely to detect. The perceived frequency range that is audible to the human ear is 20 Hz to 20 kHz. Anything that is unlikely to be detected is filtered out or converted to mono signals to take up less space.
You can see the masking tone creates a wider, masked area. Sounds that are very quiet and masked by much louder sounds are also discarded to save space.
The study of how humans perceive sound and a huge part of how lossy compression works. It is argued that our brains cannot accurately perceive every bit of data that passes our ears when listening to CD-quality audio.
Artifacts left behind by lossy compression create unwanted sounds or anomalies that are not in the original recording. These come in many different forms such as loss of bandwidth, pre-echoes, and post-echoes, double-track effect, Dynamics and phase shift and weakened low end.
Now, at kbps, MP3s filter the higher frequencies very crudely, discarding frequency content anywhere above approx. The iTunes MP3 encoder goes as far as creating distortions in this frequency range so in order to maintain full bandwidth through the iTunes MP3 encoder you must have a bit rate of kbps or higher. Even if the quiet one happens first it will be masked by the louder one if there is only a small interval of time between the two.
The masking threshold is the sound pressure level needed to make a sound audible to the human ear when in the presence of another sound known as a masker. If the sound being masked exists beyond the masking threshold then it becomes audible and we hear it as a pre or post-echo. This most often occurs with sounds from percussion instruments but is likely any shorter transient burst of noise when encoded to a format such as MP3. There is a psychoacoustic element that means one often hears the pre-echo but not the post-echo.
Forward temporal masking is much stronger than backward temporal masking which results in the post-echo being drowned out by the transient. The effect of this is most noticeably heard on vocals, creating the illusion of the voice being double-tracked. The nature of perceptual audio coding is to remove frequency content that we are unlikely to hear. The result of this can sometimes mean that our perception of the remaining frequency content can be altered.
The relative phase or timing of frequency content can be changed which can affect stereo imaging or even the transparency and clarity of the material. One of the issues the MP3 format is most known for is making a banging bassline sound timid and weak.
The amplitude of an analog signal is sampled at uniformed intervals, each sample is then quantized to the nearest value within a set range of digital steps. In a PCM stream, the amplitude of the analog signal is sampled regularly at uniform intervals, and each sample is quantized to the nearest value within a range of digital steps.
Bit depth refers to the number of possible digital values that can be used to represent each sample. Karlheinz Brandenburg , a professor at the Fraunhofer Institute was one of the lead developers of the MP3.
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