aljordan

I use a DAC that has configurable stop-band attenuation of 130 dB, 90 dB, and 50 dB. If I used an HQPlayer filter like poly-sinc that has a higher amount of stop-band attenuation than that on the FPGA, would the additional attenuation of say 130dB on the FPGA cause additional ringing, or would it do nothing because the pre-attenuation of the HQPlayer filter? I am wondering is it better to set the lowest attenuation on the FPGA when using HQPlayer, or does it not matter. Thanks

Hello, I use an Allo Piano 2.1 with Kali running on DietPi. I see that I can adjust the Piano crossover frequency starting at 60 Hz every ten Hz via the standard configuration tools shipped with the operating system. I would like to set a custom crossover frequency lower than 60 Hz. Is there any way to do this? Thank you, Alan

Thanks for the suggestion iago. I installed Voyage MPD and it works fine. Alan

Hello, I am currently running a USB stick implementation of mpdpup on an old Fit PC Slim outputting to a Berkeley Alpha USB converter. I want to keep using the Fit PC Slim as a music server, but I would like to install a Linux distribution that allows me to use the latest version of Music Player Daemon. Does anyone have suggestions of a lightweight Linux distribution, good for audio, that will run off a USB stick and allow the latest MPD version? I would rather keep it console only (no X Windows) because the Fit PC Slim is not very powerful with only 512 MB RAM. Thanks for any suggestions, Alan

Check out the following review of a budget speaker that a few people I know are very happy with. Note that I have not heard them myself, but those who own them think they sound very good. Infinity Primus P363 Loudspeaker | The Absolute Sound

Hi Ted, It has been a long time. I hope you are well. I've gone ahead and implemented reading tags from wav files (but not writing tags). If tags do not exist in the file, I am reverting to file name for title, folder name for album, and parent folder name for artist - provided that the various folders exist. Alan

Thanks all for your input. Alan

Hello, I've decided to write my own player after having some frustrations with another popular player in Windows 2012 core mode. I now have it supporting WAV, FLAC, AIFF, (and mp3). Regarding library management and searching through a collection, since WAV doesn't support metadata in any standard manner, I am wondering if people still use them and if so how do you organize your library with WAV files? Do you have a cue file in each directory that contains wav files? Do you rely on folder structure for genre, artist and album? Thanks for any information from those of you who use WAV files, Alan

On every computer I've run JRiver on, if I read music files off the network while running the Jriver convolution engine, JRiver freezes almost every time a source file sample rate changes; it freezes during the buffering stage when a new song starts. The only way I can get through this is to have JRiver resample to a single sample rate. Kind of a bummer since my main reason for using JRiver is the automatic configuration of the proper correction filter when source sample rate changes.

Thank you for this well written post. For those who have access to an equalizer in your two channel system, try playing with the area around 7 kHz a bit, and the perceived height will change as you boost and cut.

If for some reason you are still interested in trying this, I use jconvolver, jack, and zita-j2a on Linux. If you have any specific questions I could probably help. Alan

Thank you iago, Using aplay to play to a file to the device resulted in: Sample format non available Available formats: - S32_LE So then using your suggestion of format "*:32:*" fixed the issue. The working output section is audio_output { type "alsa" name "Berkeley" format "*:32:*" device "hw:1,0" } Is there a better format specifier to use for S32LE? If all this is doing is padding zeros then its fine. Thanks, Alan

Hi, I am trying to get a Berkeley Alpha USB convertor to work under Music Player Daemon but receive the following error no matter what I try: Error opening ALSA device "hw:1,0" (snd_pcm_hw_params): Input/output error Does anyone have a working audio_output configuration section for the Berkeley Alpha? I have no issues with my MPD Jack output nor a Halide Bridge. I have been unable to get Jack to work with the Berkeley either, while it works fine with my other devices. The Berkeley shows up fine when I run aplay -l, and it works fine under Windows. Thanks for any help. Alan

The Intel Ethernet drivers will not install on Windows 2012 R2 without some work. I had to follow this in order to get the LAN card to work properly: Hacking an Intel network card to work on Server 2012 R2 « FoxDeploy.com Alan

I have not tried the current JPlay two server model that seems to be popular, although in the past I have tried something similar under Linux using NetJack (which uses a networking protocol with lower overhead than TCP/IP), a powerful computer as the control PC to handle convolution needs, connected to a low power, highly optimized PC as the player. I imagine that unless one has music stored on USB drives connected to the control PC, USB would not be in the audio loop at all on that box. So my initial thought is that USB optimization would not be desired on the control PC if it optimizes available resources towards something that is not being used in the audio path. I don't know your exact configuration, but do you hear a positive difference with USB optimization enabled on the control PC as well? Thanks, Alan