3 ms·

Some of the best recordings I've heard (NPR) are only on YouTube. This leads me to believe recording quality is orders of magnitude more important than encoding

by Tarq0n 3y ago

Some of the best recordings I've heard (NPR) are only on YouTube. This leads me to believe recording quality is orders of magnitude more important than encoding, as long as a decent bitrate and encoding scheme were used.

bayindirh 3y ago
The quality ceiling for any recording you have is the quality ceiling of the weakest link in your audio pipeline. This means, to be able to get a good sound from any system, you have to feed it a good signal, and that path starts with recording. Current audio codecs are great from a psychoacoustic point of view. A good encoder can create an enjoyable file at modest bitrates (192kbps for MP3, and 128kbps for AAC IIRC), and retain most of the details. The audible residue when you subtract a MP3 from a FLAC is not details per se, but instrument separation and perceived size of the sound stage. People generally call this snake oil, but I have the same amplifier for the last 30 years, and I can say how different qualities of audio render through the same pipeline. A good recording stored losslessly can bring the concert to your home, up to a point. MP3 re-encodings of the same record will sound flatter and smaller. Lastly, it's not possible to completely contain the sound of a symphony orchestra in a stereo recording. That's not happening. So there's always a limit. If you have the time, there's a nice ABX test: http://abx.digitalfeed.net/ http://abx.digitalfeed.net/
- _joel 3y ago
  > Lastly, it's not possible to completely contain the sound of a symphony orchestra in a stereo recording Thank god we have more than 2 ears.
  - empiricus 3y ago
    Thank god they recorded the concert using binaural mics on an identical copy of your head :)
  - bayindirh 3y ago
    The problem is not the number of ears we have, but the amount of air moved by the instruments themselves and how they interact with each other. A symphony orchestra is miced per group normally (2 for violins, 2 for trumpets, etc.), but if you're around 60 people, you can mic every instrument individually. To reproduce the sound 1 to 1, you need to mic every instrument individually, and playback them with speakers matching the frequency response and air pressure . So you need speakers equal to the number and characteristics of instruments themselves. On top of that you need to record them ideal microphones and store them loslessly in the process. Otherwise, you can't create the sound by recording 100 people with 20 microphones, and downmixing them to two channels. It's not possible. I played in double bass in an orchestra, listened countless orchestras, listened the recordings of our own concerts. The gap is enormous.
    - deleted 3y ago
      [deleted]
- kjqgqkejbfefn 3y ago
  What's your take on audio systems that deliver vibrations. I have a nuraphone and a subpac. Great for listening to trap music or iranian experimental.
  - bayindirh 3y ago
    I genuinely have no opinion. I like to use a vintage amplifier with a couple of beefy bookshelf speakers. I run a couple of Heco Celan GT302s with an AKAI AM-2850. It's a very well balanced system for my needs and room size. That's a pretty nifty setup for me.
- hunter2_ 3y ago
  A reduction in soundstage/width is likely due to using "joint stereo" or "intensity stereo" encoder modes, which do things such as mid-side (M-S) conversion (which isn't itself the culprit) in order to give more bits to M (which results in better quality for sounds with high L-R correlation, like vocals) and fewer bits to S (which results in less quality for sounds with low L-R correlation, like a drum kit stereo miked). If using plain old "stereo" mode instead, this problem doesn't occur, but you need a higher overall bitrate for correlated sounds to come through at the same quality, so it's rarely used at modest bitrates and instead tends to be reserved for only the highest bitrates. Thus, comparing mp3@192 with mp3@320 often actually means comparing mp3@192joint with mp3@320stereo and therefore the listener will find very little if any improvement in the quality of mono-miked center-panned sounds (vocals, etc.) but a decent improvement in the quality of wide sounds (cymbals, reverb, string sections, etc.) since the 320 will have only a few more bits for "mid" but way more bits for "side" so to speak, relative to the 192.
  - NavinF 3y ago
    Interesting, how can I check if a file was encoded with joint stereo? I've never seen those keywords in ffprobe output Is this only an issue for mp3 or do other codecs also use fewer bits for the difference between channels?
    - bayindirh 3y ago
      "file" command outputs the encoding used, alongside details. Usage: "file test.mp3" Rsult: test.mp3: Audio file with ID3 version 2.3.0, contains: MPEG ADTS, layer III, v1, 320 kbps, 48 kHz, JntStereo
  - bayindirh 3y ago
    Thanks for the technical details, I didn't know how intensity/joint modes work, however I never use them. The tests I have done is all encoded by myself. I have purchased 24bit WAV of Radiohead's OK Computer Remastered. I encoded it to FLAC, and 320CBR Stereo with LAME. I still can feel the difference on soundstage, and can create a audible residue file by subtracting MP3 from FLAC version. I agree that current iteration of encoders create very good audio, however given that your audio system can render high resolution audio, the difference is still audible.
qingcharles 3y ago
In one of my previous lives I built an encoding platform for all the major record labels. Part of this involved listening to hundreds of tracks to try and optimize the encoder settings. It's not necessarily the quality of the original recording, but simply the type of audio. For instance, the absolute hardest, IMO, were "unplugged" albums, e.g. solo singer, acoustic guitar. Lossy compression would shit itself on those.

by Tarq0n 3y ago

bayindirh 3y ago

The quality ceiling for any recording you have is the quality ceiling of the weakest link in your audio pipeline. This means, to be able to get a good sound from any system, you have to feed it a good signal, and that path starts with recording. Current audio codecs are great from a psychoacoustic point of view. A good encoder can create an enjoyable file at modest bitrates (192kbps for MP3, and 128kbps for AAC IIRC), and retain most of the details. The audible residue when you subtract a MP3 from a FLAC is not details per se, but instrument separation and perceived size of the sound stage. People generally call this snake oil, but I have the same amplifier for the last 30 years, and I can say how different qualities of audio render through the same pipeline. A good recording stored losslessly can bring the concert to your home, up to a point. MP3 re-encodings of the same record will sound flatter and smaller. Lastly, it's not possible to completely contain the sound of a symphony orchestra in a stereo recording. That's not happening. So there's always a limit. If you have the time, there's a nice ABX test: http://abx.digitalfeed.net/ http://abx.digitalfeed.net/

_joel 3y ago
> Lastly, it's not possible to completely contain the sound of a symphony orchestra in a stereo recording Thank god we have more than 2 ears.
- empiricus 3y ago
  Thank god they recorded the concert using binaural mics on an identical copy of your head :)
- bayindirh 3y ago
  The problem is not the number of ears we have, but the amount of air moved by the instruments themselves and how they interact with each other. A symphony orchestra is miced per group normally (2 for violins, 2 for trumpets, etc.), but if you're around 60 people, you can mic every instrument individually. To reproduce the sound 1 to 1, you need to mic every instrument individually, and playback them with speakers matching the frequency response and air pressure . So you need speakers equal to the number and characteristics of instruments themselves. On top of that you need to record them ideal microphones and store them loslessly in the process. Otherwise, you can't create the sound by recording 100 people with 20 microphones, and downmixing them to two channels. It's not possible. I played in double bass in an orchestra, listened countless orchestras, listened the recordings of our own concerts. The gap is enormous.
  - deleted 3y ago
    [deleted]
kjqgqkejbfefn 3y ago
What's your take on audio systems that deliver vibrations. I have a nuraphone and a subpac. Great for listening to trap music or iranian experimental.
- bayindirh 3y ago
  I genuinely have no opinion. I like to use a vintage amplifier with a couple of beefy bookshelf speakers. I run a couple of Heco Celan GT302s with an AKAI AM-2850. It's a very well balanced system for my needs and room size. That's a pretty nifty setup for me.
hunter2_ 3y ago
A reduction in soundstage/width is likely due to using "joint stereo" or "intensity stereo" encoder modes, which do things such as mid-side (M-S) conversion (which isn't itself the culprit) in order to give more bits to M (which results in better quality for sounds with high L-R correlation, like vocals) and fewer bits to S (which results in less quality for sounds with low L-R correlation, like a drum kit stereo miked). If using plain old "stereo" mode instead, this problem doesn't occur, but you need a higher overall bitrate for correlated sounds to come through at the same quality, so it's rarely used at modest bitrates and instead tends to be reserved for only the highest bitrates. Thus, comparing mp3@192 with mp3@320 often actually means comparing mp3@192joint with mp3@320stereo and therefore the listener will find very little if any improvement in the quality of mono-miked center-panned sounds (vocals, etc.) but a decent improvement in the quality of wide sounds (cymbals, reverb, string sections, etc.) since the 320 will have only a few more bits for "mid" but way more bits for "side" so to speak, relative to the 192.
- NavinF 3y ago
  Interesting, how can I check if a file was encoded with joint stereo? I've never seen those keywords in ffprobe output Is this only an issue for mp3 or do other codecs also use fewer bits for the difference between channels?
  - bayindirh 3y ago
    "file" command outputs the encoding used, alongside details. Usage: "file test.mp3" Rsult: test.mp3: Audio file with ID3 version 2.3.0, contains: MPEG ADTS, layer III, v1, 320 kbps, 48 kHz, JntStereo
- bayindirh 3y ago
  Thanks for the technical details, I didn't know how intensity/joint modes work, however I never use them. The tests I have done is all encoded by myself. I have purchased 24bit WAV of Radiohead's OK Computer Remastered. I encoded it to FLAC, and 320CBR Stereo with LAME. I still can feel the difference on soundstage, and can create a audible residue file by subtracting MP3 from FLAC version. I agree that current iteration of encoders create very good audio, however given that your audio system can render high resolution audio, the difference is still audible.

qingcharles 3y ago

In one of my previous lives I built an encoding platform for all the major record labels. Part of this involved listening to hundreds of tracks to try and optimize the encoder settings. It's not necessarily the quality of the original recording, but simply the type of audio. For instance, the absolute hardest, IMO, were "unplugged" albums, e.g. solo singer, acoustic guitar. Lossy compression would shit itself on those.