2021-11-30, 06:37 PM
So here's my problem:
I use Kodi (Leia) as my playback device and for multichannel LPCM or FLAC files, it will send them to my receiver as 44.1khz with no resampling, same goes for AC3, DCA, TrueHD, and DTSMA. The problem I am running into is that if the .mkv is 2 channel 44.1khz, it always wants to resample and output to 48khz. I have tried all variations on the audio settings for Kodi and the ShieldTV to no avail. Rather than upgrade to a new version which breaks some other customizations I've done to the skin for cinematic playback, I'd rather look at encoding to a format that supports this sample rate. I know old school DTS wav can, and it appears TrueHD and AC3 can as well. I gave it a whirl with an old version of the Dolby Media encoder first but it keeps erroring out saying "invalid sample rate". Currently trying 640kbps AC3 which should do just fine. At what point is that mild compression worse than re-sampling it which is doing who knows what to the original audio?
In the end, I'm just trying to avoid any resampling at all on these stereo tracks so I don't introduce anything bad in the audio signal.
Of course, this could all be moot because at my age, I probably couldn't tell the different between the original wav/flac vs. a 384 kbps AC3 encode.
I use Kodi (Leia) as my playback device and for multichannel LPCM or FLAC files, it will send them to my receiver as 44.1khz with no resampling, same goes for AC3, DCA, TrueHD, and DTSMA. The problem I am running into is that if the .mkv is 2 channel 44.1khz, it always wants to resample and output to 48khz. I have tried all variations on the audio settings for Kodi and the ShieldTV to no avail. Rather than upgrade to a new version which breaks some other customizations I've done to the skin for cinematic playback, I'd rather look at encoding to a format that supports this sample rate. I know old school DTS wav can, and it appears TrueHD and AC3 can as well. I gave it a whirl with an old version of the Dolby Media encoder first but it keeps erroring out saying "invalid sample rate". Currently trying 640kbps AC3 which should do just fine. At what point is that mild compression worse than re-sampling it which is doing who knows what to the original audio?
In the end, I'm just trying to avoid any resampling at all on these stereo tracks so I don't introduce anything bad in the audio signal.
Of course, this could all be moot because at my age, I probably couldn't tell the different between the original wav/flac vs. a 384 kbps AC3 encode.