In the 21st century, everyday, more and more consumers are buying and streaming music online and parting away from buying physical CD's. In the physical world there is a standard to how CD's should be made called "Red Book Standard". Some of those specifications require that CD audio have a sample rate of 44.1kHz and a bit depth of 16 bit. The standard is the same for all CD-DA's (Compact Disk-Digital Audio) that are distributed worldwide.
- Sample Rate- A sample is the digital representation of an analog waveform, the number of samples that are captured per second is called the sample rate. The higher the sample rate, the more information of the waveform will be captured (higher quality audio).
- Bit depth - Digital representation of the amplitude (loudness) of audio waveform. The higher this resolution is, the more accurate the digital representation of the wave's amplitude is or the higher the bit depth, the more dynamic range that is captured. A 16 bit recording has a dynamic range of 96db and a 24 bit recording has a dynamic range of 144db.
We now live in a world were music is no longer just distributed physically, but now music is mostly distributed digitally online. Mastering engineers are no longer just mastering for physical release but now for digital release. The only problem with digital release is that there is not only one standard that will sound the best it can on different mediums, such as music stores, and music streaming sites. If you want your music to the sound the best it possibly can wether it's going to be released on iTunes, Radio, Soundcloud, YouTube, or simply going to end up as an mp3, then you must take a slightly different approach before exporting your final masters.
Below are a few tips on how to approach your final masters wether they are going to be released on iTunes, Radio, SoundCloud, YouTube or you simply just want the best sounding Mp3's:
Mastered for iTunes
- Final ITunes upload is a high quality ITunes Plus AAC File which uses a 44.1khz sample rate and is encoded with a 256kbs target bit rate.
- Old AAC was only 128kbs.
- iTunes wants files with more dynamic range, they are trying to bring back that 70's vinyl sound.
- Minimum upload requirement is a 24 bit lossless file (WAV/AIFF, Apple Lossless aka ALAC, FLAC, APE).
- Send at highest/original recorded sample rate.
- No Clipping (Using AURoundTripAAC Plugin).
- If one doesn’t manually properly correct this clipping, iTunes will automatically reduce the clipping of what ever file you upload thus, causing reduction in sound quality.
- Although iTunes doesn’t reject files for a specific number of clips, tracks which have audible clipping will not be badged or marketed as Mastered for iTunes.
- Apple recommends leaving -1 dB of headroom to prevent any clipping from occurring due to the noise added by the AAC encoder.
Mastered for Radio
- Sophisticated and powerful audio processing for broadcast transmission systems do not coexist well with a signal that has already been severely compressed or clipped.
- Instead of being punchy, the on-air sound produced from hypercompressed sources is small and flat, with out the dynamic range that gives music its dramatic impact.
- Broadcast processing will compress your already compressed source and will not sound better or louder on the air! It sounds more distorted, making the radio or speakers sound broken in some cases.
- Compression on top of compression will suck the drama and life out from the music.
- Use Minimal to no compression, leave the audio unsquashed. Let the broadcast processor do its work. The result will be just as loud on-air as hypercompressed material, but will have far more punch, clarity, and life.
Mastered for SoundCloud
- 24bit/192k sample rate audio files can be uploaded to sound cloud but will be transcoded to 128 kbps MP3 to prepare the audio to stream from the site.
- The higher the quality of the uploaded file, the higher quality the mp3 will be.
- You can allow users to download your original higher resolution masters or compressed mp3’s.
- If you upload an MP3, Sound Cloud will transcode it anyways, resulting in even more loss of quality by introducing more audible artifacts to audio that’s already compressed.
- SoundCloud streams such low quality audio files because they are much smaller in size, below you can see the size comparison of a 128kbit Mp3 file to higher quality wav. files.
- Set limiter margin/ceiling to around -0.3 to -1.0 dBFS to stop most of the clipping that occurs during the encoding process.
- SoundCloud does not have a feature like Apple’s SoundCheck, so an audio master destined for SoundCloud has more freedom to raise the overall RMS level for competitive loudness.
- Using a stereo imaging tool, narrow the high end between 5-20%. 128 kbps MP3 is the lowest commonly acceptable audio quality. As such, a lot of information is lost during encoding and an extremely wide mix is more susceptible to noticeable artifacts. Ironically, some narrowing can help avoid perceived loss of energy and width.
Mastered for YouTube
- YouTube transcodes all uploaded video in order to offer streaming qualities at 240p, 360p, 480p, 720, 1080p, 1440p (2k), 2160p (4k).
- Audio bitrate is not affected by video quality like in the past. The audio you hear during a YouTube video will usually be either 126 kbps AAC in an MP4 container or 155-165 kbps Opus in a WebM container (royalty-free, media file format), regardless of whether you’re playing 360p, 1080p, or any other resolution.
- Prior to 2013, YouTube played:
- 240p video with audio playback at 64kbps MP3.
- 360p and 480p video with audio playback at 128 kbps AAC.
- 720p and higher video with audio playback at 192kbps AAC.
- 24 bit 96khz audio files should be uploaded for best AAC encode.
- Mono audio files will be played at 128kbps.
- Stereo audio files audio files will be played at 384kbps.
- 5.1 audio files will be played at 512kbps.
- Set limiter margin/ceiling to around -1 dBFS.
- Not all encoders are created equal. Render from the video editor in full, uncompressed quality for both video and audio.
Mastered for Mp3
- The Mp3 compression format creates files that don’t sound exactly like the original recording because it sacrifices audio information, it is a lossy format unlike wav files which are lossless format that don’t sacrifice any audio information.
- In order to significantly make a smaller file size, Mp3 encoders have to lose audio information
- Mp3 basically uses something called “perceptual coding” to compress audio.
- Perceptual coding is a coding method that takes advantage of the human ear, screening out a certain amount of sound that it doesn’t think you can hear (elements that are masked by more important elements one can hear).
- By changing the bit rate, you can choose how much information an Mp3 file will retain or lose during the encoding and compression process (96 to 320 kb).
- Mp3 format flattens out dynamics in a song.
- At 128kbps, the encoder will remove anything at about 16kHz and above (as shown in the diagram below), so it recommended to use a low-pass filter to cut everything around and above 16kHz with a 6 or 12db per octave roll off.
- Doing this will make the important frequency information much better
- You’ll lose a little high-end (that can barely be heard), but you will gain more in mid range information.
- To retain full bandwidth (20Hz – 20kHz), Mp3 or AAC files need to be encoded at or above 256kbps.
- Avoid heavy use of saturation and distortion
- Saturation mostly affects all frequencies, the encoder will not know what parts of the distortion are intended to be “musical” and which parts could be removed).
- Avoid heavy limiting
- Leave mixes with dynamic range
- Over compressed mixes can sometimes fill in the sonic spaces that the encoder is looking for, resulting in the encoder making even more compromises and lower sound quality.
- Keep peaks below -1dBFS or even -2dBFS if you are working from a 24 bit file
- Mp3 encoders do not handle peaks that are near 0dBFS very well and the mix could end up distorting after the Mp3 encode.
- Keep RMS levels (average level) between -16 and -23dbFS.