On Sun, 24 Aug 2014, Grant wrote:
I have a USB DAC that can only handle 16/44.1 as input and output. I think ALSA will resample everything to 16/44.1 automatically, but I'd
Normally, the application connecting to ALSA looks at the port to find out what sample rates it can do and adjusts accordingly. Any recording application should lock to the interface rate with no resampleing. MP3s, Oggs and other compressed/lossy formats do internal resampling/filtering to match the desired output sampling rate anyway, but most of them are 44.1k to begin with. Wav files and flac and other no lossy formates are the only ones where resampling is needed if they are not already 44.1k. In general any wav files will be ones you recorded and already be the right sample rate.
I think what I am saying is that for most cases the sample rate of your audio IF doesn't matter. So adding resampling to everything doesn't make sense... maybe try without first.
like that to happen with the highest quality resampler which I think is samplerate_best. I use xfce4 and I don't want to install pulseaudio. I've added the following to /etc/asound.conf: defaults.pcm.rate_converter "samplerate_best" Should that do it? How can I verify that it's working?
Playback a 48K wav file. if there are no extra clicks or pops it is fine. -- Len Ovens www.ovenwerks.net _______________________________________________ Linux-audio-user mailing list Linux-audio-user@xxxxxxxxxxxxxxxxxxxx http://lists.linuxaudio.org/listinfo/linux-audio-user
