[Alsa-devel] BUG REPORT: NFORCE2/Soundstorm recording broke
My setup is: ALSA driver 0.9.7a, 0.9.7 lib Linux kernel 2.4.21 RedHat 9 ASUS A7N8X Deluxe/ NFORCE2 chipset with Soundstorm onboard sound Playback of both PCM and MIDI sound works fine but the problem is that recording doesn't work under ALSA. When I type 'arecord -l' I get this: List of CAPTURE Hardware Devices card 0: nForce2 [NVidia nForce2], device 0: Intel ICH [NVidia nForce2] Subdevices: 1/1 Subdevice #0: subdevice #0 but the when I issue 'arecord -t wav -f cd -d 6 test.wav' I just get a silent test.wav file. I've tried recording under Audacity too and that produces similiar results, unless I use the OSS kernel driver and then I can record in 16-bit, 44.1khz stereo but not mono and no full-duplex either. The funny thing is that full duplex recording under ALSA did actually work under Audacity with my same hardware setup- this was when I was running Mandrake 9.1 and an earlier ALSA version (0.9.2 or thereabouts), so I know it is possible. I've reported this bug to this list before but either it just got ignored or there are no ALSA developers who own a NFORCE2 board. If this isn't the right place to report such bugs I would appreciate if someone forwarded this to the right address or told me where to send it. Thanks, dan http://www.transelement.co.uk __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] .asoundrc plugin setup
i have some troubles with an usb-audio device (an imic) on linux ppc
(an ibook2), i think becouse of endianness related troubles..
my big trouble is that jack says:
[EMAIL PROTECTED]:~$ jackd -d alsa -d hw:1
jackd 0.80.0
[...]
JACK compiled with System V SHM support
loading driver ..
creating alsa driver ... hw:1|hw:1|1024|2|48000|nomon|swmeter|rt
open
Sorry. The audio interface "hw:1"doesn't support either of the two
hardware sample formats that jack can use.
ALSA: cannot configure capture channel
starting engine
engine driver not set; cannot start
[...]
i've setupped an .asounrc like that:
pcm.imic {
type hw
card 1
}
ctl.imic2 {
type hw
card 1
}
pcm_slave.sl3 {
pcm imic
format S16_LE
}
pcm.imic2 {
type plug
slave sl3
}
and jack at least start, but it dont playback anything:
[EMAIL PROTECTED]:~$ jackd -d alsa -d imic2
jackd 0.80.0
[...]
creating alsa driver ... imic2|imic2|1024|2|48000|nomon|swmeter|rt
open
You appear to be using the ALSA software "plug" layer, probably
[...]
You appear to be using the ALSA software "plug" layer, probably
[...] starting engine
ALSA lib pcm_hw.c:494:(snd_pcm_hw_start) SNDRV_PCM_IOCTL_START failed:
Broken pipe
could not start playback (Broken pipe)
jackd: signal 2 received
jack main caught signal 2
received signal 15 during shutdown (ignored)
[EMAIL PROTECTED]:~$
its a mine fault or alsa fault?? i think that with that .asoundrc this
things should work...
tanks for attention
wil
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[Alsa-devel] Mixer functions
Hi, With 5.1 channel sound being used quite a lot now, I was wondering what to do about volume controls. For example, I have: - Front speakers controlled by PCM and Master slider. Rear speakers controlled by Surround slider Center controlled by Center Slider LFE controlled by LFE slider. I would assume that a "Master" slider would control all 6 channels at the same time. So, in this setup, the Master slider is not actually acting as a Master, but instead, only a Front speaker control. I suggest that in these cases, "Master" gets renamed to something else, and then alsa-lib takes over providing a "Master" alsa-lib could then also use config files, to define the mixer elements. We could then provide mixer elements that do proper "Master", "Front/Rear Fader" etc. I know you said earlier that you are in the process of developing a better mixer, possibly having 0db points etc. Could you add these suggestions, so that they are possible as well. Another problem I have come across, is that some sound cards have a headphone amplifier on the front speaker output, and some sound cards have a headphone amplifier on the rear/surround outputs. E.g. ALC650 has headphone on front output. AD1985 has headphone on surround (but is switchable to front). Does anyone know if the AC97 standard is headphones on front, or headphones on rear/surround ? I was talking to a user who has sound coming out of a speaker output, but not the headphone or center/lfe outputs, also the slider name controlling the speaker output was "surround". It would be nice to find out from yourselves what the AC97 standard says it should be, and I will then post a PATCH to get the AD1985 to act in the same way as all other sound cards. Cheers James --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ xine-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/xine-devel --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] SB live goes directly to DD mode
Hello, I have tried kernel 2.6.0-test6 with a SB live and a DD receiver connected to the SPDIF. As soon as ALSA initialize, the receiver goes to DD mode and I don't hear anything as long as I don't sent ac3 (for example in playing a DVD). Is this a new feature, could I disable it? Thank you very much, Grégoire http://magma.epfl.ch/greg ICQ:16624071 mailto:[EMAIL PROTECTED] --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] SP/DIF input of Soundblaster Live 5.1
Hi there, I'm trying to route spdif signal from my dvb card through my soundcard, a sb live 5.1, to my amplifier. I connected the spdif out of the dvb card to the "CD SPDIF" connector of the sb live card. Is this the proper way to do it, ie, should it work or could it be made to work? I've played around with the settings in alsamixer, but so far been unable to hear anything from my amplifier. The most relevant switches seem to be "IEC985 Coaxial Capture", "IEC985 Optical Capture" and "IEC985 Optical Raw [on/off]". Is this a supported feature in the alsa driver? I'm using version 0.9.7 of the driver. Any help greatly appreciated. Best regards, Juha --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Scatter-Gather buffer allocation before running on HDSP-MADI ?
Hello ! (A question from an newbie-developer to make it perfect ;-) I am writing an ALSA-lowleveldriver for the RME HDSP-MADI card. (which have 64Audion in and out and an 128in64(= 8192Fader) Mixer see http://www.rme-audio.de/hdsp/hdspmadi.htm ) Coding is quite complete but I have trouble with the memory management. The Card uses ScatterGather Buffer, each channel 64kB and this has to be asigned (to be safe) before activation, so I want to allocate SGbuffer (64+64)*64*1024=8388608 in 4k-blocks. I do a snd_pcm_lib_preallocate_pci_pages_for_all(hdspm->pci, pcm,8388608l,8388608l) after making the pcm_device. Here the questions: 1) How can I get the sgbuf pointer ? ... since in the streams runtime is not set and therefore (besides the Documentation says is (snd_pcm_sgbuf_t*)substream->dma_private which I couldnt find) sgbuf = (snd_pcm_sgbuf_t*)substream->runtime->dma_private; is not assigned. but when I activate the card (for MIDI for example) there must be memory asigned und I think doing it in hw_params is to late. 2) If it is bound to substream is always the same substream the capture or playback or are the assigned dynamically on calling ? mfg winfried ritsch --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Unexpected hw_pointer value
hello, i have an ES1983S Maestro-3i on a c600 dell laptop i got this message in syslog : ALSA sound/core/pcm_lib.c:214: Unexpected hw_pointer value (stream = 0, delta: -944, max jitter = 1024): wrong interrupt acknowledge? what does it mean ? --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Unexpected hw_pointer value
Hi, On Sat, Oct 04, 2003 at 11:12:00PM +0200, Jon wrote: > hello, > > i have an ES1983S Maestro-3i on a c600 dell laptop > i got this message in syslog : > > ALSA sound/core/pcm_lib.c:214: Unexpected hw_pointer value (stream = 0, > delta: -944, max jitter = 1024): wrong interrupt acknowledge? I get a lot of these as well with the in-development Aureal driver. I'm also curious as to what they mean. (I seem to be the only person having the problem). They mostly occur while playing music in xmms with xmms alsa plugin. They fill up syslog with hundreds of megabytes. :) Thanks, -- Ryan Underwood, , icq=10317253 --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
