Hello,
my movie has the following audio data (from SD-TV):
Format : MPEG Audio
Format version : Version 1
Format profile : Layer 2
Codec ID : 3
Duration : 22mn 45s
Bit rate mode : Constant
Bit rate : 192 Kbps
Channel(s) : 2 channels
Sampling rate : 48.0 KHz
Compression mode : Lossy
Delay relative to video : -98ms
Stream size : 31.3 MiB (4%)
Now, when I encode that mp2(!), which is 192kbps to AAC, AC3 or MP3 at 192kbps, is this just a "format change" or is it loosing quality?
Like, when I take 1/4 of 1/4 of a cake I of course have less, than in the beginning. The example is problematic, since the bitrate, of course will result in the same file size, but will it result in the same file quality, too?
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There has to be a loss of quality since you are starting with a lossy format.
The definition of 'Quality Loss' means that data is lost in the encoding and different codecs will do it differently. Even if, at the end, it sounds the same to your ears.
But why are you re-encoding ? Mp2 audio is supported for Pal-DVD and AFAIK is fine for other container formats such as mp4. -
Regarding handling things differently from codec to codec (i.e. for example cutting away different frequencies and making use of other psycho-acoustic tools, to put it in a simple example for me), would I maintain somehow keeping more of the mp2 deails, if I use a higher bitrate for AAC/AC3/MP3, so that from one codec to the other less stuff in a certain area, that e.g. AAC will handle different, will be cut?
I understand, that I can't make better quality, if I e.g. encode a 192kbps MP3 to a 320kbps MP3, of course, but since e.g. AAC focuses on different areas, as you said, would the higher bitrate help?
It seems Handbrake won't let me do an auto-passthrough of the mp2, hence I have to re-encode. I would rather like to keep the mp2, but Avidemux is to complex for when using x.264 for the video track and I think mpegstreamclip doesn't have a good x.264 codec (though it supports passthrough of mp2). There are no more Apps I know of for Mac.Last edited by MovingParts; 23rd Mar 2016 at 15:25.
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I am no audiophile but I take video as my example here.
Higher bitrate only helps if you re-encode into an intermediate uncompressed codec - for audio PCM is one - before re-encoding with the other lossy codec. Then your re-encode can be at a higher bitrate than the original mp2.
That might help to 'preserve' the quality. But I can not vouch for that.Last edited by DB83; 23rd Mar 2016 at 16:28. Reason: clarity
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More bitrate = less quality lost. So encoding 192 kbps mp2 to 320 kbps mp3 has better quality than 192 kbps mp2 to 192 kbps mp3. But: many modern encoders are very good and most people will not be able do hear a difference. Just because there's a theoretical difference does not mean you will actually hear one.
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DB83: Lossy transcoding (of both video and audio) always* goes through an intermediate uncompressed stage due to the decompression step that must happen in order to feed the compressed input to the second encoder. Typically this is invisible to the user.
* Possible exception: Re-encoding the same codec with specialized tools. I think DVD Shrink had a magical mode that could utilize the existing motion vectors and just re-quantize to create a lower bitrate.
OP: Highest quality outputs for what you are doing, in order:
Original MP2 stream (most compatible, depending on container)
Uncompressed PCM (same quality as above)
Lossless (FLAC, ALAC, etc.) (same quality as above, smaller, less compatible for hardware players)
High-bitrate lossy transcode
Low-bitrate lossy transcode
But then there's the law of diminishing returns, already alluded to. -
Tanks for the additional answers!
As you all said, I probably won't be able to tell a difference, but I might just try, at least now I know, that it is not totally stupid to even try it.
Btw. "law of diminishing returns" was what I wanted to say with my cake example, but couldn't phrase it right. -
Meanwhile I found out some of my source movies, have three tracks, 2 being mp2 at 256kbps and 1 being AC3 at 448kbps. So with these files I can just passthrough the AC3 track (Handbrake has a passthrough function that only works for AC3).
However theoretically I wonder, if a 256kbps mp2 converted to 256kbps whatever, would be worse than a 448k AC3 converted to 256k MP3 or AAC. -
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While I would certainly use that, when I need it to be really small, I guess I would rather use AAC at a higher bitrate or does HE-AAC compare to lets say a 256-320kbps AAC, when HE-AAC is at 80kbps?
Also, is HE-AAC not optimised for speech rather than movie sound with music? -
While I would certainly use that, when I need it to be really small, I guess I would rather use AAC at a higher bitrate or does HE-AAC compare to lets say a 256-320kbps AAC, when HE-AAC is at 80kbps?
Also, is HE-AAC not optimised for speech rather than movie sound with music? -
HE-AAC uses spectral band replication, which discards some of the higher frequency stuff then kind of "fakes it" on playback. It saves a "side track" for restoring it at half the sample rate of the main audio. In theory if you play HE-AAC using a player that only supports AAC-LC, it'll work, but the high frequencies will be missing. Perceived quality
I generally re-encode audio as AAC using the default VBR quality for QAAC or NeroAAC, and for 5.1ch audio that ends up in the vicinity of 400kbps or more (the DTS movie audio I just re-encoded resulted in 462kbps). As a result I generally don't re-encode 5.1ch or 2ch AC3 because there's not enough of a bitrate reduction to make it worthwhile, but for DTS or lossless formats there is.
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