>> 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. I think you're saying that the ALSA resampler won't be used if the upstream application does the resampling itself. Is that correct? How can I find out if ALSA is the one resampling in a particular scenario? - Grant _______________________________________________ Linux-audio-user mailing list Linux-audio-user@xxxxxxxxxxxxxxxxxxxx http://lists.linuxaudio.org/listinfo/linux-audio-user
