Hi, I have a number of 24bit 96kHz files. My hardware platform is an Nvidia Ion based system and I am connecting to an external DAC via SPDIF. The Ion platform claims to be using a realtek ALC662 device to output over SPDIF. Looking at the 662 datasheet, it says the following "The ALC662 series support 16/20/24-bit SPDIF output function and a sampling rate of up to 96kHz". In xbmc, I have selected the custom audio output and the value is set to "hw:0,1" (the SPDIF port). Turning debugging on in xbmc and the system sees the file as 24/96. WAVCodec::Init - Sample Rate: 96000, Bits Per Sample: 24, Channels: 2 but the ALSA initialise initialises to 16 bit: CALSADirectSound::CALSADirectSound - Channels: 2 - SampleRate: 96000 - SampleBit: 16 - Resample false - Codec - IsMusic true - IsPassthrough false - audioDevice: hw:0,1 Looking at the hardware params: root@xbmc:~# cat /proc/asound/card0/pcm1p/sub0/hw_params access: RW_INTERLEAVED format: S16_LE subformat: STD channels: 2 rate: 96000 (96000/1) period_size: 4096 buffer_size: 16384 The question is: what happens to the 24bit audio? Am I getting full 24bit out of the SPDIF port or is it being down converted to 16bit? If it isn't 24bit, is there any way for me to force it to use 24 bit (the device is S16_LE and S32_LE capable but surprisingly doesn't support S24_LE). Is this a bug in xbmc? I'm running ALSA v 1.0.20 but I can't find any mention of related changes here in the upgrade logs on the ALSA wiki. ------------------------------------------------------------------------------ ThinkGeek and WIRED's GeekDad team up for the Ultimate GeekDad Father's Day Giveaway. ONE MASSIVE PRIZE to the lucky parental unit. See the prize list and enter to win: http://p.sf.net/sfu/thinkgeek-promo _______________________________________________ Alsa-user mailing list Alsa-user@xxxxxxxxxxxxxxxxxxxxx https://lists.sourceforge.net/lists/listinfo/alsa-user
