- I am trying to playback a 320kbps mp3 audio file at its recorded quality. If I were to drag and drop it into audacity, would it playback at the 320kbps bitrate by default? If not, how should I change it so it plays back at 320 kbps?
- I also have another mp3 that is 128 kbps. would it play back at 128 kbps with the optimal settings for 320 kbps?
You get the “actual quality” of the MP3.
When you open a file in Audacity, or play it in any software it gets decompressed and converted to “regular” PCM (like a CD or WAV file). This is what goes to your DAC (digital-to-analog converter).
The sample rate (kHz) is unchanged.
Audacity works in 32-bit floating point so the MP3 is decompressed to floating-point. MP3 actually uses a kind of floating point (but it doesn’t store individual samples). Of course if the original was 16-bit CD audio, the sound quality doesn’t get “better”.
Then when you play it, the bit depth is converted to 16 or 24-bits to match your DAC. (That happens with any file in any format with any player software.)
Now… If you open an MP3 in Audacity (or any regular audio editor) and then you re-export as MP3, you are going through another generation of lossy compression and some “damage” accumulates. You may not notice any quality loss but it’s something you should be aware of, and you should try to minimize the number if times it’s re-compressed.
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Thank you for the response,
As someone with no audio engineering education I have another question. If i go into Edit > Preferences > Quality, i see sections for Real-Time Conversion and High-Quality conversion. What do these mean and do they have any effect on playback quality? I adjusted the Real Time Conversion Sample Rate Converter to “Best Quality (Slowest)” as opposed to “Medium Quality” like it is by default thinking that it would make the audio playback better quality, but when I reopened audacity the Real Time Conversion Sample Rate Converter was set back to Medium Quality
As a practical matter those settings are not important unless you are making 8-bit files.
Real time is when you are playing in Audacity. High Quality is when you export to a new file.
It’s related to dither. Dither is added noise that’s supposed to sound better than quantization noise.
You can hear quantization noise at 8-bits but under normal listening conditions you can’t hear quantization noise or dither at 16-bits or higher.
The “rule” is to dither when you down-sample. If you are re-exporting as MP3 you’re not down-sampling the bit depth so you can set it to “none”. If you happen to know the original was 24-bits before compressing to MP3, and you are exporting to 16-bit WAV, you should dither.
I missed that part of your question. You can increase the bitrate but you can’t increase the MP3 “quality”. MP3 makes the file smaller mostly by throwing-away information. It’s gone and it’s not coming back. The lower the bitrate, the smaller the file and the more information has been thrown-away. It’s “smart” and it tries to throw-away little sounds that you can’t hear because they are masked (drowned-out) by other louder sounds that are close in frequency.
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So would changing the Real Time Conversion Sample Rate Converter to Best Quality make the audio play back at a better quality?
I guess what im asking is if I have an mp3 at 320 kbps and an mp3 at 128 kbps, will they both individually play back at their most optimal quality (320 kbps and 128 kbps) in the player?
I should clarify that when i said “increase” i was referring to the quality that audacity was able to play the audio at and not necessarily the audio itself. For example, if audacity could only play 128 kbps by default in the timeline player, and there was an option to increase that to 320 kbps and beyond.
I have 2 downloaded mp3s, one at 320 kbps and the other at 128 kbps. Is the audacity player/timeline capable of playing 320 kbps audio in its original bitrate by default, or are there settings that would have to be changed to make it possible?
Possibly, if you are down-sampling. But you may not know. Most MP3s were ripped from CDs (16-bits) and most modern DACs/soundcards are 24-bits so you’re usually not down-sampling. And since Audacity is not intended as and everyday audio player, it’s usually not critical. And, like I said it’s usually not critical anyway and you probably won’t hear any difference either way. This is mostly “theory”.
You’ll get the actual MP3 quality. The data is thrown-away during compression when the MP3 was made. No further damage happens during decompression. The decompressed bitrate is higher.
The bitrate for CD quality audio (16-bits, 44.1kHz) is 1411 kbps (16 bits x 44.1K x 2 channels = 1411kbps). With Audacity’s internal 32-bit format, it’s twice that.
Specifically the mp3 audio is from videos i downloaded from youtube. Does that change your answer to the first question? Should I play it safe by putting the Real Time Conversion Sample Rate Converter to best quality, or could that negatively effect audio quality?
So you’re saying that I can play the downloaded audio back on audacity at the same quality/bitrate that I would get if i watched it on youtube?
Feel free to use Audacity with default settings. It won’t get worse))
By the way, the sound of videos from YouTube is not encoded in mp3. If you have sound in mp3, it means that you downloaded it from some service that re-encoded the sound. That is, your sound is already compressed twice.
The Real Time Conversion Sample Rate Converter only affects the quality of the audio being played by Audacity (the sound you hear when you click the play button in Audacity). It doesn’t affect the quality of the imported audio, and it doesn’t affect the quality of exported audio.
Yep, most audio from YouTube is in a more modern AAC or Opus format. Opus at 128k (format ID 251) is going to be just about the best quality you can get from most YT videos. Opus is roughly as good as MP3 at double the bit rate, so 128k Opus is roughly the same as 256k MP3.
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