What Kind of Files Can You Upload to Soundcloud
While some streaming services like Amazon Music Hard disk drive and Tidal are now offer lossless audio, many others like Spotify, Apple tree Music, and SoundCloud yet apply lossy audio compression techniques to deliver music. Of those, SoundCloud has always been unique in how easy it makes instant uploads for creators.
Perhaps information technology's due to that very ease that questions similar, "Why does my music sound dissimilar on SoundCloud?" or "What can I do to make my music sound better on SoundCloud?" seem to come up upward more than often than they do for other streaming services.
Despite SoundCloud introducing a new "mastering" feature to optimize streaming playback, knowing what actually happens to your audio during streaming and mastering is primal to understanding how to produce a track with the highest possible sound quality for streaming. So let's take a look at why those sonic changes occur, and what nosotros tin do to minimize them.
In this piece you'll acquire:
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How to optimize your songs for streaming on SoundCloud and other compressed audio formats
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What you can and tin't control in the procedure
The bottom line
To get to the bottom of this, I prepared 40 masters of a single song—20 at 44.1 kHz and 20 at 48 kHz—and uploaded them all to SoundCloud. For each sample rate, I methodically varied the parameters of elevation level, crest factor, frequency-specific width, and total width. I then played them all back off SoundCloud, recording the output bitstream pre-conversion—once more at 44.1 kHz and 48 kHz—for analysis and comparison confronting the originals. This yielded a whopping eighty versions of the song!
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20 uploaded and recorded at 44.1 kHz
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xx uploaded at 48 kHz and recorded at 44.1 kHz
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xx uploaded at 44.1 kHz and recorded at 48 kHz
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20 uploaded and recorded at 48 kHz
Testing forty versions of a song
Afterwards level matching them all for a fair comparison, I got to work listening and measuring to decide which factors played the biggest role in preserving—or degrading—audio quality during format conversion and streaming playback. At the cease of the day the parameter which fabricated the biggest impact was: width! Non just that, but all the other variables had little to no impact (caveats alee).
To understand why this is, how you can potentially have advantage of information technology, and why you might not desire to worry nigh it at all, read on!
Manipulating width for a "meliorate" encode
I should qualify what I hateful by "better." Actually, what nosotros're talking almost is an encode which is perceptually closer to the source. Even so, the steps nosotros're taking to get at that place involve making some sacrifices to the source. So while the encode and the source may audio more alike, the cumulative difference between the encode, the source, and what yous were originally trying to achieve may still be fairly noticeable.
That qualifier aside, here are a few things you can practice to minimize the differences between the source and the encode:
Narrow the high-finish
Using a tool like the Imager in Ozone 9, try narrowing frequencies above about 8 kHz. I can't requite you a precise corporeality, as it will very much depend on the amount of width that y'all had in that range to begin with. Effort soloing that band and reducing the width until it occupies about half of the stereo field between your speakers. This will help reduce some of the high-frequency washiness that is so common with low-bitrate lossy codecs.
Narrow mid and low frequencies
If you desire, and your master tin can handle it, try narrowing the mid and low bands as well. Try setting the mid band to well-nigh 1–8 kHz, and the low ring below 1 kHz. You could even split this into two ranges: 400–thousand Hz and below 400 Hz. Yous'll likely want to leave the mid—and low-mid if you're using information technology—bands adequately shut to their original width, however, y'all may be able to get away with narrowing lower frequencies a bit more than. Any footling bit helps.
Use a mono primary
This is absolutely an extreme solution, just if you lot can justify it, a mono source will requite you the "all-time" encode—again, meaning perceptually closest to the source, albeit at present in mono. This is because you lot're essentially request the encoder to exercise half equally much work by encoding a single channel. In plough, this means the encoder tin can allocate it's entire bandwidth to that 1 channel, rather than having to divide it between two channels.
The reasons width plays such a critical role in encoder performance are hugely circuitous, only can be summarized as follows: virtually lossy encoders similar AAC, MP3, and Opus employ a technique known every bit articulation stereo encoding. This means that rather than encoding both left and correct channels independently, they employ multiple techniques such as mid/side and intensity-stereo coding to optimize bandwidth allocation to where it will be most noticeable—often the center of the stereo image.
The cease result is that ultra-wide stereo signals ofttimes suffer from quality deposition more noticeably than practice narrower ones. Additionally, loftier frequencies require more bandwidth to encode. Thus, by reducing the width of high frequencies, not only practice you gratis upwardly some bandwidth for the encoder, allowing it to allocate its $.25 more efficiently, merely you also preclude some of the more than noticeable, warbly, washy distortion from showing up in the encode.
A great way to experiment with the furnishings of these changes in real-fourth dimension is past using the Codec Preview in Ozone 9 Avant-garde. Try using MP3 at 128 kbps or AAC at 256 kbps—two of the common codecs used by SoundCloud depending on the playback platform and subscription level—and tweaking Imager parameters. Yous can even use the "Solo Artifacts" role to hear how changes in width affect the underlying distortion added by the codec.
Codec Preview in Ozone ix
All the other $.25
I would be remiss if I didn't address things like summit level, crest-factor, and file format for upload, and then let's talk about those at least a little.
In all my recent tests, height level did not take a noticeable impact on encoder performance—at least not direct. By this, I mean that so long every bit at that place wasn't whatsoever clipping, the encoder performance between versions with different amounts of superlative headroom was identical.
However, because lower bitrates—such every bit those often used past SoundCloud—can cause peak level overshoot of a decibel or more, it'southward good practice to set the ceiling of your limiter to -1 or -1.5 dB and apply a True Peak limiter such equally the Ozone Maximizer. This helps prevent clipping on playback, especially through cheaper consumer devices.
The story with crest factor is largely the same. While information technology doesn't accept a direct, dramatic bear upon on encoder performance, a lower crest factor will often result in higher superlative level overshoot—something which ultimately often results in DAC clipping and distortion. This has the slightly ironic consequence of requiring additional peak headroom—or a lower limiter ceiling—the higher you push your boilerplate level, something which can quickly plow into a losing battle.
This is another area where Codec Preview in Ozone 9 Advanced can exist enormously helpful. By turning on Detect "True Peaks" in the I/O options and listening through the MP3 128 kbps codec, yous tin can fine-tune the Maximizer threshold and ceiling to attain an optimal level while fugitive postal service encode clipping.
Checking mail-encode peak headroom in Ozone 9
As for upload format, the official recommendation from SoundCloud is a sixteen-fleck, 48 kHz WAV file. This reason for this is that of the several codecs used, the bulk of them are set to have in a 48 kHz file, so this minimizes the amount of sample rate conversion that will take place.
That said, sample rate conversion has become extremely transparent, and in my tests neither the upload nor playback sample rates had an observable effect on encoder performance or playback quality.
The one caveat here is that if yous enable downloads on SoundCloud, the file y'all upload is the 1 your fans get when they download. Thus, if you desire them to receive a 320kbps MP3, that's what yous'll demand to upload. However, this results in transcoding from i lossy format to another, which never sounds peculiarly good.
In curt, if yous want the best streaming quality possible, upload a xvi-bit WAV at 44.i or 48 kHz. If, on the other hand, you want to enable downloads, upload the file you want your fans to receive, but know that if it'southward a lossy file, streaming quality volition suffer. Since these days downloading a local copy is probably not as common as it one time was, this may be a moot indicate.
Conclusion
To wrap up I desire to consider a few reasons why perhaps you shouldn't worry too much about all the factors we've just discussed.
Kickoff and foremost, SoundCloud may well update the codecs they use in the future simply as they have in the past. When that happens they volition re-encode all uploaded music to accept advantage of the new codec(southward). It's for this very reason that they themselves urge creators not to effort to optimize files too much for a specific codec.
Second, while yous can control the width, sample rate, etc. of the file you upload, you lot can't control how your fans will listen to it. Of grade, this is true of the vast bulk of playback mediums. It bears repeating here though because even on SoundCloud alone, the playback experience can vary depending on subscription level and playback device. Consider carefully whether it'southward worth sacrificing some of the width and spaciousness of your track merely for the lowest common denominator.
Hopefully, this has armed you non but with some of the tools to improve encoder performance when uploading to SoundCloud simply also the wisdom to know when, when non, and how strongly to wield them. Good luck, and happy mastering!
Source: https://www.izotope.com/en/learn/mastering-for-compressed-audio-formats.html
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