I created a test file exactly like yours using ffmpeg and a test file
from 2L.
When playing the file to a SB player which is capabale of 48kHz - LMS
creates a trasncoding chain as follows:
Code:
--------------------
"/xx/7.9/latest/trunk/Bin/x86_64-linux/faad" -q -w -f 1
"/xx/albums/TestFiles/Test2L38-48kHz-1.m4a" |
"/xx/7.9/latest/trunk/Bin/x86_64-linux/sox" -q -t wav - -t flac -C 0
--------------------
This shows
1. Transcoding from ALAC to WAV is done by faad.
2. LMS showing files as 16 bit should not affect output as faad reads
file directly (i.e. LMS does NOT read the file)
3. faad converts the ALAC 24 bit file into WAV 24bit (s24le) stream.
4. sox is used to compress WAV stream into Flac stream to be sent to
player. As a test when the flac stream is saved as a file it is Flac
48kHz/24bit in s32le format.
The data stream sent to the SB-Receiver is a Flac / 24bits
I cannot see any truncation anywhere in the chain.
Except for the LMS displaying file as 16bit - what else indicates
truncation ?
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