So floating point MP3 decoding producing floating point samples just above full scale (0.527 dB) sounds plausible. However it makes sense to me for Audacity to delay that quantization step until you tell it what to do with the file - to avoid over-processing the audio. In such a case the floating point output would get usually quantized to behave like a standard MP3 decoder. ![]() That's plausible because I have seen documents which tell me the MP3 decoding process can be implemented in floating point. I assume the Audacity's MP3 decoder is written to produce 32-bit floating point samples. ![]() Here is a hypothesis that someone could check. When asked to export the file I see you can tell Audacity what resolution to quantize (16-bit PCM, 24-bit PCM) or to use 32-bit float, etc. At least that's what my copy of Audacity tells me when I import an MP3 file. I use Audacity fairly rarely but I think Audacity works internally with 32-bit floating point for audio samples. In the above example, the mp3 file was only +0.527dB above maximum level, but I've seen examples that are almost +2dB above max.ġ) Given that the peaks of the waveform are no longer clipped once the level has been reduced, am I correct to infer that the clipping was NOT baked-in to the mp3 file when it was created but has instead been caused by Audacity when it opened the file?Ģ) Is this behaviour unique to Audacity? In other words, if I play the same mp3 file in iTunes or stream it over Spotify, will the peak levels stay below -0dBFS?ĩ9.9% of my library is lossless so the problem, if it exists, will have negligible impact on me, but I'm thinking of the implications this has on Spotify users etc. The following screenshots illustrate what I'm trying to describe: Applying this inverse value reduces the level of the waveform to below -0dBFS, and running clipping analysis confirms that there are no longer any instances of clipping. Interestingly, however, when I highlight said waveform and click 'Amplify' - a feature that allows you to increase or decrease the level of the waveform - the field is populated by the inverse of the number of dB the waveform is above -0dBFS. Running clipping analysis confirms that there is clipping throughout the track. ![]() When I open an mp3 file in Audacity, the waveform is often hard up against the -0dBFS ceiling and appears to be clipped. I also tried changing the normal volume target in mp3gain higher to about 98.0db it makes all the songs louder but get clipping in almost every song.This might be one for our resident analysis boffin Audiomisc. ![]() When i apply max no clip gain for each file i get different db for every song from 98.0 to 89.0 then i go to itunes raise to volume of every song to about 90% but some still sound lower then other what should i do to make them all the same volume without clipping or is clipping really noticable I have no clue what to do to make all my songs really loud Some songs are in aac format and mp3 format and different kbps. Should the bitrate for each song be the same and what type or does it matter when getting the songs louder with no clipping. Would that make all the songs about the same volume and how high should i raise the volume in itunes to make the volume all the same and loud after apply max no clip gain for each file. Then go to itunes and select all the songs and increase the volume. I put all my songs on Mp3Gain then select apply max no clip gain for each file. I want all my songs in itunes to be loud and all the same volume without any clipping.
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