Extract core with eac3to and tsmuxer spectrum different

[tebasuna51]

Is not so simple:

"... joint intensity coding and sum/difference coding may be employed to further improve audio quality. The
optional LFE channel is compressed by: low-pass filtering, decimation and mid-tread scalar quantization."

The LFE channel without high frequencies need only a low bitrate to be encoded.
And some channels can be joined to optimize compression, and front channels can have preference for high fequencies over back channels...

[filler56789]

Originally Posted by theliver9x

Let me ask more, dts 1509kbps 6 channels, so each internal channel will have a bitrate is 251kbps?

tebasuna51's answer explained it well, it's not that simple.
But if one wants to simplify anyway [but not too much], one can say that 1509 kbps for 5.1 DTS is (roughly) equivalent to 300 kbps per channel (the ".1" channel not being counted, of course).

Which is overkill, granted. DTS-CD uses 1234.8 kbps @ 44.1 kHz, which is the same as 1344 kbps @ 48 kHz...
Fortunately the BD-specs do allow lower bitrates for 5.1 DTS audio.
Unfortunately the average BD production is lazy and stupid

[theliver9x]

Originally Posted by tebasuna51

Is not so simple:

"... joint intensity coding and sum/difference coding may be employed to further improve audio quality. The
optional LFE channel is compressed by: low-pass filtering, decimation and mid-tread scalar quantization."

The LFE channel without high frequencies need only a low bitrate to be encoded.
And some channels can be joined to optimize compression, and front channels can have preference for high fequencies over back channels...

Thanks. And when convert audio, i see a message saying "remapping channel", so what is remapping and it have change the raw audio data or not?

[tebasuna51]

The channel order inside a compressed stream can have a different order in dts, ac3, aac, etc.
Any decoder output always (remapping) the standard wav (M$) file channel order, any encoder must accept the standard wav order like input.

All is automatic, you don't need do nothing.

[theliver9x]

Originally Posted by tebasuna51

The channel order inside a compressed stream can have a different order in dts, ac3, aac, etc.
Any decoder output always (remapping) the standard wav (M$) file channel order, any encoder must accept the standard wav order like input.

All is automatic, you don't need do nothing.

I convert the wav file to ac3 with eac3to, length of ac3 file output is 21ms longer than the original wav, why is it different and what do i have to do to get the ac3 file of the same length as the original wav file?

[tebasuna51]

You always get a AC3 file duration multiple of 32 ms (if 48 KHz) because a AC3 stream is a set of frames of that length.

The encoders delay the wav file with 256 silence samples (5.33 ms at 48 KHz) and fill the las frame with silence samples until the 32 ms full frame.