In general, the weight of the audio is not a problem; and the 128kbps quality is enough for everything [until here I understand everything well]... But my question is, when a bitrate is chosen in the GUI, what audio bitrate are we referring to?
example: in MP3 the bitrates are
8 Kbps Mono: Telephone Sound. | 16 Kbps Mono: Better quality than shortwave. | 32 Kbps Mono: Better quality than AM.
64 Kbps Stereo: Better quality than FM. | 112 - 128 Kbps: Near CD quality. | 160 Kbps: Quality closer to CD. |192 Kbps: Practically CD quality. |256 Kbps: CD quality practically indistinguishable from an original CD. | 320 Kbps: CD quality.
However, perceptually, an AAC at 128 contains much more range than an MP3 128 (an AAC at 192 is considered equivalent to 256 Mp3) and in turn a Vorbis that can even implement up to 500 kbps, managing to surpass both aac and mp3 especially below 128.
Therefore, when a bit rate value is choiced in the GUI, it is based on a specific fixed scale, to the chosen format
in codec, or is it just a symbolic bitrate? [CBR/VBR apply here?]
I usually only record google Meet conferences so I use 64 kbps (based on the mp3 quality table, but using the vorbis codec).
maybe it's a bit trivial but I occupy SSR in daily meetings of 2 to 3 hours on average (around 6 hours a day in different shots) so the idea of being able to save some bitrate in space and not "oversample" an audio already compressed by itself and in telephone quality/background hiss-noise I am quite struck by it
or eventually the opposite, granting audio quality when required
indirectly, the bitrate has something to do with the classic pulseaudio "too many audio samples" message. ?
In general, the weight of the audio is not a problem; and the 128kbps quality is enough for everything [until here I understand everything well]... But my question is, when a bitrate is chosen in the GUI, what audio bitrate are we referring to?
example: in MP3 the bitrates are 8 Kbps Mono: Telephone Sound. | 16 Kbps Mono: Better quality than shortwave. | 32 Kbps Mono: Better quality than AM. 64 Kbps Stereo: Better quality than FM. | 112 - 128 Kbps: Near CD quality. | 160 Kbps: Quality closer to CD. |192 Kbps: Practically CD quality. |256 Kbps: CD quality practically indistinguishable from an original CD. | 320 Kbps: CD quality.
However, perceptually, an AAC at 128 contains much more range than an MP3 128 (an AAC at 192 is considered equivalent to 256 Mp3) and in turn a Vorbis that can even implement up to 500 kbps, managing to surpass both aac and mp3 especially below 128.
Therefore, when a bit rate value is choiced in the GUI, it is based on a specific fixed scale, to the chosen format in codec, or is it just a symbolic bitrate? [CBR/VBR apply here?]
I usually only record google Meet conferences so I use 64 kbps (based on the mp3 quality table, but using the vorbis codec). maybe it's a bit trivial but I occupy SSR in daily meetings of 2 to 3 hours on average (around 6 hours a day in different shots) so the idea of being able to save some bitrate in space and not "oversample" an audio already compressed by itself and in telephone quality/background hiss-noise I am quite struck by it or eventually the opposite, granting audio quality when required indirectly, the bitrate has something to do with the classic pulseaudio "too many audio samples" message. ?