How to Change How Many Hertz Your Uploading Music
While some streaming services like Amazon Music HD and Tidal are now offering lossless audio, many others like Spotify, Apple tree Music, and SoundCloud all the same employ lossy audio pinch techniques to deliver music. Of those, SoundCloud has always been unique in how piece of cake it makes instant uploads for creators.
Perhaps it's due to that very ease that questions similar, "Why does my music audio unlike on SoundCloud?" or "What tin can I do to brand my music sound better on SoundCloud?" seem to come more than frequently 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'south take a look at why those sonic changes occur, and what we can do to minimize them.
In this piece you'll larn:
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How to optimize your songs for streaming on SoundCloud and other compressed sound formats
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What you tin can and tin can't control in the process
The bottom line
To get to the bottom of this, I prepared 40 masters of a unmarried song—20 at 44.one kHz and 20 at 48 kHz—and uploaded them all to SoundCloud. For each sample rate, I methodically varied the parameters of peak level, crest factor, frequency-specific width, and full width. I so played them all back off SoundCloud, recording the output bitstream pre-conversion—once again at 44.1 kHz and 48 kHz—for analysis and comparison confronting the originals. This yielded a whopping lxxx versions of the song!
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20 uploaded and recorded at 44.one kHz
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20 uploaded at 48 kHz and recorded at 44.1 kHz
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20 uploaded at 44.1 kHz and recorded at 48 kHz
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20 uploaded and recorded at 48 kHz
Testing twoscore versions of a song
After level matching them all for a fair comparison, I got to work listening and measuring to determine which factors played the biggest office in preserving—or degrading—sound quality during format conversion and streaming playback. At the finish of the day the parameter which made the biggest impact was: width! Not merely that, only all the other variables had little to no touch (caveats ahead).
To understand why this is, how you tin potentially accept reward of it, and why you lot might not want to worry about it at all, read on!
Manipulating width for a "better" encode
I should qualify what I mean by "ameliorate." Really, what we're talking well-nigh is an encode which is perceptually closer to the source. However, the steps nosotros're taking to get there involve making some sacrifices to the source. Then while the encode and the source may sound more than alike, the cumulative divergence between the encode, the source, and what you were originally trying to achieve may all the same be fairly noticeable.
That qualifier aside, here are a few things y'all can do to minimize the differences betwixt the source and the encode:
Narrow the loftier-end
Using a tool like the Imager in Ozone 9, endeavor narrowing frequencies above about 8 kHz. I can't give you a precise amount, equally it volition very much depend on the amount of width that y'all had in that range to brainstorm with. Try soloing that band and reducing the width until information technology occupies nigh half of the stereo field betwixt your speakers. This will aid reduce some of the high-frequency washiness that is so common with low-bitrate lossy codecs.
Narrow mid and low frequencies
If you want, and your master can handle it, effort narrowing the mid and low bands as well. Try setting the mid band to virtually 1–viii kHz, and the low ring below one kHz. You could even carve up this into two ranges: 400–yard Hz and below 400 Hz. You'll likely want to leave the mid—and low-mid if you lot're using information technology—bands fairly shut to their original width, notwithstanding, you may exist able to become away with narrowing lower frequencies a fleck more. Any little scrap helps.
Use a mono master
This is absolutely an extreme solution, but if you tin can justify it, a mono source volition give you the "best" encode—again, meaning perceptually closest to the source, albeit now in mono. This is considering you're essentially asking the encoder to do half every bit much work by encoding a single channel. In turn, this ways the encoder can allocate it's entire bandwidth to that one channel, rather than having to divide it between two channels.
The reasons width plays such a disquisitional role in encoder performance are hugely circuitous, but tin be summarized every bit follows: near lossy encoders similar AAC, MP3, and Opus employ a technique known every bit joint stereo encoding. This means that rather than encoding both left and right channels independently, they employ multiple techniques such every bit mid/side and intensity-stereo coding to optimize bandwidth allocation to where information technology will be most noticeable—often the centre of the stereo epitome.
The end result is that ultra-broad stereo signals often suffer from quality deposition more noticeably than do narrower ones. Additionally, high frequencies require more bandwidth to encode. Thus, by reducing the width of high frequencies, not only do you free upward some bandwidth for the encoder, allowing it to allocate its bits more than efficiently, only you besides preclude some of the more noticeable, warbly, washy distortion from showing upwardly in the encode.
A peachy style to experiment with the effects of these changes in existent-time is by using the Codec Preview in Ozone nine Avant-garde. Try using MP3 at 128 kbps or AAC at 256 kbps—2 of the common codecs used by SoundCloud depending on the playback platform and subscription level—and tweaking Imager parameters. You tin fifty-fifty use the "Solo Artifacts" function to hear how changes in width affect the underlying distortion added by the codec.
Codec Preview in Ozone 9
All the other bits
I would be remiss if I didn't address things similar peak level, crest-cistron, and file format for upload, so allow's talk about those at least a picayune.
In all my contempo tests, peak level did not accept a noticeable impact on encoder functioning—at least not directly. By this, I mean that so long as at that place wasn't any clipping, the encoder performance between versions with unlike amounts of peak headroom was identical.
Nonetheless, because lower bitrates—such as those often used by SoundCloud—can cause elevation level overshoot of a decibel or more than, it'due south expert practice to set the ceiling of your limiter to -ane or -1.v dB and utilise a True Peak limiter such as 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 have a direct, dramatic affect on encoder performance, a lower crest factor will often upshot in college peak level overshoot—something which ultimately often results in DAC clipping and baloney. This has the slightly ironic consequence of requiring boosted peak headroom—or a lower limiter ceiling—the college you button your average level, something which can chop-chop plough into a losing battle.
This is some other area where Codec Preview in Ozone ix Advanced can exist enormously helpful. By turning on Notice "True Peaks" in the I/O options and listening through the MP3 128 kbps codec, you can fine-tune the Maximizer threshold and ceiling to achieve an optimal level while fugitive mail encode clipping.
Checking mail-encode peak headroom in Ozone nine
As for upload format, the official recommendation from SoundCloud is a sixteen-chip, 48 kHz WAV file. This reason for this is that of the several codecs used, the majority of them are set to take in a 48 kHz file, so this minimizes the amount of sample rate conversion that will take identify.
That said, sample charge per unit conversion has go extremely transparent, and in my tests neither the upload nor playback sample rates had an observable effect on encoder operation or playback quality.
The one caveat here is that if you enable downloads on SoundCloud, the file you upload is the one your fans get when they download. Thus, if you want them to receive a 320kbps MP3, that'southward what you'll need to upload. Nevertheless, this results in transcoding from i lossy format to another, which never sounds peculiarly good.
In brusque, if y'all want the best streaming quality possible, upload a 16-fleck WAV at 44.1 or 48 kHz. If, on the other hand, yous want to enable downloads, upload the file y'all want your fans to receive, but know that if it'south a lossy file, streaming quality volition suffer. Since these days downloading a local copy is probably not as common as it once was, this may be a moot point.
Conclusion
To wrap up I want to consider a few reasons why perhaps yous shouldn't worry likewise much nigh all the factors we've merely discussed.
First and foremost, SoundCloud may well update the codecs they use in the future merely as they have in the past. When that happens they volition re-encode all uploaded music to have reward of the new codec(south). Information technology's for this very reason that they themselves urge creators not to attempt to optimize files too much for a specific codec.
Second, while y'all tin control the width, sample rate, etc. of the file yous upload, yous can't command how your fans will heed to it. Of course, this is truthful of the vast majority 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 simply for the lowest common denominator.
Hopefully, this has armed yous not simply with some of the tools to improve encoder performance when uploading to SoundCloud but also the wisdom to know when, when not, and how strongly to wield them. Proficient luck, and happy mastering!
Source: https://www.izotope.com/en/learn/mastering-for-compressed-audio-formats.html
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