On Mon, 31 Aug 2015 09:58:24 -0600 KI7MT <ki7m...@gmail.com> wrote: > To check the available sampling rates, he can use lsusb to find the > device ID, then look at the sample frequencies, mine is by TI, here an > example, in a terminal: > > * lsusb > * My Device --> Bus 002 Device 004: ID 08bb:2906 Texas Instruments > > * Then probe the ID for tSamFreq: > * lsusb -v -d 08bb:2906 | grep tSamFreq | sed -e "s,^.* ,," | sort -nu > > My Results: > 8000 > 11025 > 16000 > 22050 > 32000 > 44100 > 48000 > > > 73's > Greg, KI7MT
I very much doubt it is a sampling rate problem, as the Rigblaster is in use with WSJT modes already. If the sampling rate was not available there would have been very loud noises made by users, and seen on the lists. I would strongly suggest Haaken has a look at page 10 of the Rigblaster user manual, maybe more than a look, sorta read , learn and inwardly digest. Looking at the documentation, this is very much a user interface problem and not a wsjt problem. > > > On 08/31/2015 09:36 AM, Bill Somerville wrote: > > On 29/08/2015 14:26, Håken Hveem wrote: > >> Audio Input Output Device Name > >> Device Channels Channels > >> ------------------------------------------------------------------ > >> 0 2 8 HDA Intel PCH: ALC892 Analog (hw:0,0) > >> 1 0 2 HDA Intel PCH: ALC892 Digital (hw:0,1) > >> 2 2 0 HDA Intel PCH: ALC892 Alt Analog (hw:0,2) > >> 3 0 8 HDA NVidia: HDMI 0 (hw:1,3) > >> 4 0 8 HDA NVidia: HDMI 1 (hw:1,7) > >> 5 0 8 HDA NVidia: HDMI 2 (hw:1,8) > >> 6 0 8 HDA NVidia: HDMI 3 (hw:1,9) > >> 7 1 2 RIGblaster Advantage Audio: USB Audio > >> (hw:2,0) > >> 8 128 128 sysdefault > >> 9 0 8 front > >> 10 0 8 surround40 > >> 11 0 128 surround41 > >> 12 0 128 surround50 > >> 13 0 8 surround51 > >> 14 0 8 surround71 > >> 15 0 2 iec958 > >> 16 0 2 spdif > >> 17 32 32 pulse > >> 18 0 2 dmix > >> 19 32 32 default > >> > >> User requested devices: Input = 7 Output = 7 > >> Default devices: Input = 19 Output = 19 > >> Will open devices: Input = 7 Output = 7 > >> ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear > >> ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM > >> cards.pcm.center_lfe > >> ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side > >> bt_audio_service_open: connect() failed: Connection refused (111) > >> bt_audio_service_open: connect() failed: Connection refused (111) > >> bt_audio_service_open: connect() failed: Connection refused (111) > >> bt_audio_service_open: connect() failed: Connection refused (111) > >> ALSA lib pcm_params.c:2162:(snd1_pcm_hw_refine_slave) Slave PCM not usable > >> ALSA lib pcm_params.c:2162:(snd1_pcm_hw_refine_slave) Slave PCM not usable > >> ALSA lib pcm_params.c:2162:(snd1_pcm_hw_refine_slave) Slave PCM not usable > >> Expression 'alsa_snd_pcm_hw_params_set_rate_near( pcm, hwParams, > >> &setRate, NULL )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 3201 > >> ALSA lib pcm_params.c:2162:(snd1_pcm_hw_refine_slave) Slave PCM not usable > >> ALSA lib pcm_params.c:2162:(snd1_pcm_hw_refine_slave) Slave PCM not usable > >> ALSA lib pcm_params.c:2162:(snd1_pcm_hw_refine_slave) Slave PCM not usable > >> ALSA lib pcm_params.c:2162:(snd1_pcm_hw_refine_slave) Slave PCM not usable > >> Cannot connect to server socket err = No such file or directory > >> Cannot connect to server request channel > >> jack server is not running or cannot be started > >> Expression 'parameters->channelCount <= maxChans' failed in > >> 'src/hostapi/alsa/pa_linux_alsa.c', line: 1514 > >> Expression 'ValidateParameters( inputParameters, hostApi, > >> StreamDirection_In )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', > >> line: 1838 > >> Error opening input audio stream. > >> Errortext: Invalid number of channels > >> Number: -9998 > >> Error opening output audio stream. > >> Errortext: Invalid sample rate > >> Number: -9997 > >> Error 3 in JTaudio, you will only be able to work offline. > > Hi All, > > > > watching the slow progress of this thread in terms of resolutions has > > annoyed me enough to look at the source code of WSJT. Although I am not > > an expert in Linux audio configurations, maybe the following comments > > might help: > > > > WSJT opens portaudio input and output streams expecting to get a 16bit > > 11.025kHz sample rate and two channels. It would appear, from the error > > messages, that the RigBlaster codec is mono for input and doesn't > > directly support 11.025kHz sample rates. > > > > If I were having this issue I would investigate how to create a output > > virtual sound device which supports a 11.025kHz sample rate and an input > > virtual sound device that maps a single channel to left and right stereo > > along with a 11.025kHz sample rate. This may be possible using ALSA > > configuration (~/.asoundrc) or maybe by using pulseaudio which has > > advanced virtual device construction and an ALSA backend capability so > > could be accessed via the portaudio library that WSJT uses. > > > > The way WSJT is coded does not allow for other sample rates or channel > > configurations so the above mapping are needed for it to work with the > > RigBlaster codec. > > > > YMMV > > 73 > > Bill > > G4WJS. > > > > ------------------------------------------------------------------------------ > > _______________________________________________ > > wsjt-devel mailing list > > wsjt-devel@lists.sourceforge.net > > https://lists.sourceforge.net/lists/listinfo/wsjt-devel > > > -- -- Best wishes /73 Richard Bown Email : rich...@g8jvm.com HTTP : http://www.g8jvm.com nil carborundum a illegitemis ################################################################################## Ham Call: G8JVM . 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