Aaron, I am not running pulseaudio or jackd. As Rafael mentioned I changed "channel = 2" and now DarkIce no longer errors out when starting so that put me on a better path than where I have been for the last few days. However there is no audio being captured. I tried changing the device to hw:0,0 hw:0,2 and default without any success. I added the line to create a local dump file to hear if anything was being captured. It sounds like a 60Hz hum with no other audio on all three files - hw00, hw02 and default. root@atomd2500cce:~# aplay -vv ./hw00_dump.mp3 Playing raw data './hw00_dump.mp3' : Unsigned 8 bit, Rate 8000 Hz, Mono Plug PCM: Rate conversion PCM (48000, sformat=U8) Converter: linear-interpolation Protocol version: 10002 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : U8 subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 8 buffer_size : 2730 period_size : 170 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 170 period_event : 0 start_threshold : 2730 stop_threshold : 2730 silence_threshold: 0 silence_size : 0 boundary : 768426686420090880 Slave: Route conversion PCM (sformat=S32_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : U8 subformat : STD channels : 1 rate : 48000 exact rate : 48000 (48000/1) msbits : 8 buffer_size : 16384 period_size : 1024 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 Slave: Soft volume PCM Control: PCM Playback Volume min_dB: -51 max_dB: 0 resolution: 256 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 16384 period_size : 1024 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 16384 period_size : 1024 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 4611686018427387904 Hardware PCM card 0 'HDA Intel' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 16384 period_size : 1024 period_time : 21333 tstamp_mode : ENABLE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1 stop_threshold : 4611686018427387904 silence_threshold: 0 silence_size : 4611686018427387904 boundary : 4611686018427387904 appl_ptr : 0 hw_ptr : 5120 ##################################################+| MAX Thanks, -Brian From: darkice-bounce@xxxxxxxxxxxxx [mailto:darkice-bounce@xxxxxxxxxxxxx] On Behalf Of AaronHorn . Sent: Tuesday, February 11, 2014 8:32 PM To: darkice@xxxxxxxxxxxxx Subject: [darkice] Re: DarkIce, ALSA & Intel HDA Audio This seems like an ALSA issue. Out of interest have you tried using it in stereo? Also is anything else using ALSA e.g. pulseaudio or Jackd? On 12 February 2014 01:20, <brian_f_cox@xxxxxxxxxxx> wrote: Has anyone had luck running DarkIce and ALSA using an Intel HDA onboard audio? I recently purchased an Intel Desktop Board D2500CCE running Slackware64-14.1 and want to move IceCast/DarkIce off an old system running Slackware64-14.0. On the old system IceCast/DarkIce and the onboard ASUS/VIA audio work flawlessly. I figured this would be a simple move however it is not. I keep getting the error: "DarkIce: AlsaDspSource.cpp:173: can't set channels [1]" on the new Intel system. After spending a few days searching the Internet I am not having much success finding a resolution. I am leaning towards the soundcard as I have reformatted the drive and tried Slackware-14.0, Slackware-14.1 and Slackware64-14.1. I am able to pipe audio in through the line-in and get output through the speakers. I also am able to play a WAV file so I know the onboard sound is working. I checked the onboard audio settings in the BIOS and only options are to Enable to Disable the on-board sound. I have recompiled DarkIce-1.0, DarkIce-1.2, faac, lame, libogg, libvorbis and two-lame packages under each OS with both DarkIce-1.0 and DarkIce-1.2 presenting the same error: "DarkIce: AlsaDspSource.cpp:173: can't set channels [1]". The darkice.cfg file is the same copy that works fine on the old Slackware64-14.0 ASUS system. root@atomd2500cce:~/darkice# /usr/bin/darkice -c /etc/icecast/darkice.cfg DarkIce 1.2 live audio streamer, http://code.google.com/p/darkice/ Copyright (c) 2000-2007, Tyrell Hungary, http://tyrell.hu/ Copyright (c) 2008-2013, Akos Maroy and Rafael Diniz This is free software, and you are welcome to redistribute it under the terms of The GNU General Public License version 3 or any later version. Using config file: /etc/icecast/darkice.cfg Using ALSA DSP input device: hw:0,0 Using POSIX real-time scheduling, priority 4 DarkIce: AlsaDspSource.cpp:173: can't set channels [1] root@atomd2500cce:~/darkice# arecord -l **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC888-VD Analog [ALC888-VD Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 2: ALC888-VD Analog [ALC888-VD Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 root@atomd2500cce:~/darkice# cat /etc/icecast/darkice.cfg # this section describes general aspects of the live streaming session [general] duration = 0 # duration of encoding, in seconds. 0 means forever bufferSecs = 10 # size of internal slip buffer, in seconds #reconnect = yes # reconnect to the server(s) if disconnected # this section describes the audio input that will be streamed [input] device = hw:0,0 # OSS DSP soundcard device for the audio input sampleRate = 22050 # sample rate in Hz. try 11025, 22050 or 44100 bitsPerSample = 16 # bits per sample. try 16 channel = 1 # channels. 1 = mono, 2 = stereo # this section describes a streaming connection to an IceCast2 server # there may be up to 8 of these sections, named [icecast2-0] ... [icecast2-7] # these can be mixed with [icecast-x] and [shoutcast-x] sections [icecast2-0] bitrateMode = cbr # constant bit rate format = mp3 # format of the stream: mp3 bitrate = 16 # bitrate of the stream sent to the server server = localhost # host name of the server port = 8000 # port of the IceCast2 server, usually 8000 password = password # source password to the IceCast2 server mountPoint = mystream # mount point of this stream on the IceCast2 server name = myaudio # name of the stream genre = generic # genre of the stream public = yes # advertise this stream? I have also tried changing the darkice.cfg device to hw:0,2. Neither /dev/snd nor /dev/dsp as the device work and produce different errors. Does anyone have any pointers for something simple I may be missing? Thank you in advance for any assistance anyone can provide. -- Regards, Aaron Horn, aaronhorn@xxxxxxxxx.