On 01/23/2017 07:44 PM, chris hermansen wrote: > Sean and list > > > > On Jan 23, 2017 16:06, "sean darcy" <seandar...@gmail.com > <mailto:seandar...@gmail.com>> wrote: > > alsa-lib-1.1.1 > > aplay -l > **** List of PLAYBACK Hardware Devices **** > card 0: DragonFly [AudioQuest DragonFly], device 0: USB Audio [USB > Audio] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > ............. > > [video@sixcore]$ aplay -D hw:0,0 -v ~/2-15_Route_66.wav > Playing WAVE '/home/video/2-15_Route_66.wav' : Signed 16 bit Little > Endian, Rate 44100 Hz, Stereo > aplay: set_params:1297: Sample format non available > Available formats: > - S24_3LE > > Any help appreciated. > > > Your DragonFly looks like mine, only accepts 24 bit word length. > > Use plughw rather than he. > >
Thanks for the quick repsonse. Now I can't get alsa to use the sync resampler. aplay -v -D plughw:1,0 2-15_Route_66.wav Playing WAVE '2-15_Route_66.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Plug PCM: Linear conversion PCM (S24_3LE) ............ even though: cat asound.conf # # Place your global alsa-lib configuration here... # defaults.pcm.rate_converter "samplerate_best" and ls /usr/lib64/alsa-lib/ libasound_module_conf_pulse.so libasound_module_ctl_oss.so libasound_module_ctl_pulse.so libasound_module_pcm_oss.so libasound_module_pcm_pulse.so libasound_module_rate_samplerate_best.so libasound_module_rate_samplerate_linear.so libasound_module_rate_samplerate_medium.so libasound_module_rate_samplerate_order.so libasound_module_rate_samplerate.so smixer What am I missing this time ? sean ------------------------------------------------------------------------------ Check out the vibrant tech community on one of the world's most engaging tech sites, SlashDot.org! http://sdm.link/slashdot _______________________________________________ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
