> If you wish to support downsampling - you should look at main > convert.conf file to how to do downsampling using flac.
Downsampling is not needed in my case; all of the players support 24/192/2ch. > Sorry for the dumb question, but the audio source is only available in > analogue format ? Other than that, would creating a source stream using > (eg) vlc work ? that source stream could then be a radio feed fed back > for LMS - if you are not sensitive to delay, that would probably work, I > think Yes, the audio from my HQ audio sources is only available in analog format. While the streaming with vlc would also (hopefully) work too, I wouldn't be the only one using this solution, and I wanted to have a nice interface and an easy-to-control method as well as synchronization of the playback streams. I have a feeling that would be complex with that direct streaming method. > Remove the PCM line as there is no easy way this is going to work. > Remove the MP3 because even though it can be made to work - why waste > time working out the correct lame option settings. > After this suggestion, I wanted to make sure that whatever command I have in the custom-convert.conf file would be more or less usable from the command line. I did several tests with this, and ran into problems. I tested playback of the .flac files I created with mplayer, since arecord/aplay doesn't support flac files. I downloaded a 24/192/2ch flac test file from Linn Recordings to ensure that I could playback a 24/192/2ch file locally on my server unit, which I was able to do through mplayer with no problems. However, when I tried to create a line close to what I'd be using in the custom-convert.conf file using "arecord -d0 -c2 -f S24_LE -r 192000 -traw -D hw:0 | flac -cs --endian=little --sign=signed --channels=2 --bps=24 --sample-rate=192000 --compression-level-0 - -o flac_raw24.flac", I simply got static with warped music underneath. When I change the bits per sample (in both the arecord portion and the flac portion) though, mplayer plays back the file fine. These two behaviors also occur if I change the -t field to wav. Changing the sample rate doesn't change the behavior. The mplayer states that the file is s32le format (which I thought might be a problem), but it also states that the working flac file from Linn is s32le. Also, I tried to simply record 24 bit and 16 bit audio to a wav file (not using the flac conversion) and play it back with mplayer. mplayer will sound the same as the 24 bit flac files I created (static, warped music), but when I play it through aplay, it plays fine. Does anyone know why the flac conversion is crapping out like this at 24 bits? The Wolfson card can definitely support 24/192/2ch capture/playback. I know this thread is running a bit off topic, but I feel like this is the last hurdle before a working solution. I've attached screenshots of all the flac conversions/wav captures/playbacks I attempted and the mplayer results when I play them. test192.flac is the Linn test file: 16968 16969 16970 16971 16972 Thanks for all the responses so far! Cheers +-------------------------------------------------------------------+ |Filename: mplayer Output 4.png | |Download: http://forums.slimdevices.com/attachment.php?attachmentid=16972| +-------------------------------------------------------------------+ ------------------------------------------------------------------------ mike_b16's Profile: http://forums.slimdevices.com/member.php?userid=63606 View this thread: http://forums.slimdevices.com/showthread.php?t=102689 _______________________________________________ plugins mailing list plugins@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/plugins
