Some questions on transcoding

[Lex]

Hello Simon,

I hope you could clarify a few things for me. My situation is that I am using REW-created IR convolution files together with transcoding:

- After convulution the volume is about 8 dB lower than before. In my version of ffmpeg the gtype=gn function still works, but I found that in some cases the replay gain of the stream is actually increased.
I have tested a file that originally that had peak value of 1.00000, which, after convolution had a higher replaygain. This makes me wonder whether the gtype=gn command could result in clipping. Would it not be better to use another command to increase to volume with a fixed conventional gain of, say, 2?

- Whatever setting I choose for DSF to PCM conversion, the sound is always affected. Since I want the room correction also for DSF, this is an issue (as my Linn DS3 natively handles DSD). I asked Gemini, who came with a quite interesting recommendation which I would want to try. Unfortunately, I don't know how to enter this in mminimstreamer.
The idea is that DSD produces lots of HF noise, and we would want to include a low pass filter at 28 kHz before transcoding to PCM. The suggested PCM sample frequency is 88 kHz. A restoration of the volume with 6 dB is also suggested (although I would prefer a factor of 2 over 6 dB since that would avoid rounding errors).

- Apparently you can set the precision of the afir filter to "double". Would that yield more accurate results?

Thanks in advance for your response!

[simoncn]

MinimStreamer transcoding works by running FFmpeg with filter options provided by the user in the stream.options property. These options are very complex and I don't have the advanced knowledge of FFmpeg and transcoding that would be needed to answer your questions.

It is possible that someone else with more knowledge of FFmpeg might respond to your post here. Your other option is to direct these questions to the FFmpeg user community or developer community. To enable this, I suggest you try running FFmpeg as a standalone command using a small number of sample audio files that illustrate the problems you are having. If you are able with help from the FFmpeg community to find the correct FFmpeg options to make this standalone FFmpeg transcoding work correctly, I will do my best to help you to apply these options to MinimStreamer.

[Lex]

Hello Simon,

I tried your suggestion and spent way too much time trying different settings. But, whatever settings I use, it does not seem possible to retain the full audio quality converting DSD to PCM.
I tried all target sample frequencies (88, 176 and 352 kHz), different lowpass filters (4th order LR at 40 kHz would be my preference, made sure I use sox at 32 bits precision, dithering etc.

It occurred to me that perhaps my DAC is better at playing DSD than it is at playing PCM, but the CD versions also sound better than the converted SACDs, in some ways. Sure, the CD is not as refined and the bass is all over the place (relatively speaking), but some of the space and transient edge of percussion etc. seems to get lost in the conversion process.

I believe I must accept this, and make a choice if I prefer the room correction, or the unrestrained sound quality of the DSD file. At least: I won't look into this any further.

[simoncn]

This was my experience as well. I have purchased the same music in DSD and PCM and untranscoded DSD was clearly superior to PCM or transcoded DSD. This was true even when comparing untranscoded DSD with untranscoded hi-res PCM (24/192).