Room correction with convolution per sample rate

[simoncn]

Thanks very much for testing the new build and for your very helpful feedback.

When you set dsf:wav24;176 in stream.transcode and also enable convolution, the following is happening:

1) The DSD audio data is converted to PCM with a sample rate of 352800 Hz. This value is fixed and does not depend on the transcoding output sample rate in stream.transcode.

2) The output from step 1 is convolved using the 352800 Hz convolution filter file (or the closest sample rate available if this file is not found).

3) The output from step 2 is downsampled to 176400 Hz as specified by your stream.transcode setting.

This means that MinimStreamer is correctly using the 352800 Hz convolution filter file and your DAC is correctly reporting the sample rate of the stream that it receives as being 176400 Hz.

For comparision, the following is happening for a PCM audio stream:

1) Not applicable

2) The output from step 1 is convolved using the convolution filter file that exactly matches the sample rate of the PCM audio stream (or the closest sample rate available if this file is not found).

3) The output from step 2 is downsampled or upsampled if this is specified by your stream.transcode setting.

In step 2 (for both DSF and PCM input files), if MinimStreamer provides a convolution filter file that doesn't exactly match the sample rate of the audio data, FFmpeg automatically resamples the convolution filter file to the sample rate of the audio data, just like JRiver.

Because of this, I don't think MinimStreamer should prevent convolution if the convolution input file and audio stream don't have matching sample rates, as this would negate a useful feature of FFmpeg.

To prevent convolving 352800 Hz and 384000 Hz files because of the stuttering issue, you could use an input type filter in stream.transcode to prevent these sample rates from being transcoded. This assumes that your DAC can accept DSD audio input.