[Alsa-user] ALSA-only apps sharing hardware with PulseAudio
My use-case is as follows: I have a headless PC (a Raspberry Pi) that I want to use as a generic sound server. I want to run three audio applications on this; two of those apps will talk to ALSA directly, and one needs to use Pulseaudio. I have only one actual hardware audio interface, a USB DAC. So all three apps need to share the same hardware. Also, the USB DAC provides hardware-based volume control; I would like all apps (regardless of being ALSA- or PulseAudio-based) to also share that control. My understanding is that there appears to be two different ways to do this: (1) Configure the dmix plugin for ALSA, and have all applications, including PulseAudio, use dmix (rather than any one app directly taking exclusive control of the hardware device) (2) Configure a "pulse" virtual ALSA device that allows ALSA apps to use the ALSA API as a passthrough to PulseAudio And maybe there are other ways? My question is: is one way better than the other? Or, maybe a better question, what are the pros and cons of each approach? It feels like the second option is more typically recommended, but I find that counter-intuitive, since if the application is written against the ALSA API, why does PulseAudio need to be involved at all? The app pushes data to ALSA, which forwards to PulseAudio, which ultimately pushes it back to ALSA. A related question: is there any way to maintain "bitperfect" playback with any of these device-sharing schemes? That is, if there is only one application using the device, then the PCM data will be passed directly to the DAC, without any resampling. Only if the hardware does not support the sample rate, or if multiple applications want to output sound simultaneously, will the PCM data be changed from the source. Thanks! Matt ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] query alsa for supported sample rates and formats?
On Fri, Apr 17, 2009 at 10:14:16AM +0200, Clemens Ladisch wrote: > Matt Garman wrote: > > Is there a way to query alsa to see what sample rates and > > formats the sound hardware natively supports? > > Try the attached program. I get the following output when I run this for my M-Audio Audiophile 24/96. No surprises: what led me to asking this question was my observation that I couldn't play CD audio (S16_LE) using hw, and have to use dmix or plughw. Device: hw (type: HW) Access types: MMAP_INTERLEAVED RW_INTERLEAVED Formats: S32_LE Channels: 10 Sample rates: 8000 11025 16000 22050 32000 44100 48000 64000 88200 96000 Interrupt interval: 20-409500 us Buffer size: 20-819125 us Now, before Clemens was nice enough to post that program, I posed the same question on the M-Audio.com forums: http://forums.m-audio.com/showthread.php?p=38192 The response seems to suggest that the card should support more formats. Or perhaps I'm misreading the responses (or the tech that answered the question doesn't really understand my question). Thoughts? Thanks again, Matt -- The NEW KODAK i700 Series Scanners deliver under ANY circumstances! Your production scanning environment may not be a perfect world - but thanks to Kodak, there's a perfect scanner to get the job done! With the NEW KODAK i700 Series Scanner you'll get full speed at 300 dpi even with all image processing features enabled. http://p.sf.net/sfu/kodak-com ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] ICE1712 (Delta 1010) sample rate issue
On Fri, May 08, 2009 at 02:12:55PM +, Antony Gelberg wrote:
> I've re-installed a Delta 1010 (ice1712) in this machine, and got
> it working with jack, which is fine for DAW stuff, but the more
> simple task of getting music played through it via alsa eludes me.
>
> I have no .asoundrc at present, my hardware info is at
> http://www.alsa-project.org/db/?f=f0c76123a68f7596dbc339fea5a92b101f70f2cf,
> an example error is at http://pastebin.ca/1416214. Basically it's something
> like:
> $ alsaplayer -o alsa -d hw:2,0
> error on set_format SND_PCM_FORMAT_S16
>
> I've googled the "error on set_format" but no joy. Every program
> I try to use to play music with via hw:2,0 complains about the
> sample rate in some way.
My hunch is that your soundcard doesn't directly support that format
(signed 16 bit). I have the M-Audio Audiophile 24/96 card, which
also uses the ice1712 driver, and it looks like it only supports
signed 32 bit data.
Try compiling and running the attached program. It was posted a few
weeks ago by Clemens Ladisch, in response to me asking if there was
a way to determine what sound formats are supported by the hardware.
See:
http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg24774.html
If indeed that is the case, I think the only solution is to use dmix
or plughw (instead of hw directly) to convert the data to a format
that is natively supported by your hardware.
-Matt
/*
* hw_params.c - print hardware capabilities
*
* compile with: gcc -o hw_params hw_params.c -lasound
*/
#include
#include
#define ARRAY_SIZE(a) (sizeof(a) / sizeof *(a))
static const snd_pcm_access_t accesses[] = {
SND_PCM_ACCESS_MMAP_INTERLEAVED,
SND_PCM_ACCESS_MMAP_NONINTERLEAVED,
SND_PCM_ACCESS_MMAP_COMPLEX,
SND_PCM_ACCESS_RW_INTERLEAVED,
SND_PCM_ACCESS_RW_NONINTERLEAVED,
};
static const snd_pcm_format_t formats[] = {
SND_PCM_FORMAT_S8,
SND_PCM_FORMAT_U8,
SND_PCM_FORMAT_S16_LE,
SND_PCM_FORMAT_S16_BE,
SND_PCM_FORMAT_U16_LE,
SND_PCM_FORMAT_U16_BE,
SND_PCM_FORMAT_S24_LE,
SND_PCM_FORMAT_S24_BE,
SND_PCM_FORMAT_U24_LE,
SND_PCM_FORMAT_U24_BE,
SND_PCM_FORMAT_S32_LE,
SND_PCM_FORMAT_S32_BE,
SND_PCM_FORMAT_U32_LE,
SND_PCM_FORMAT_U32_BE,
SND_PCM_FORMAT_FLOAT_LE,
SND_PCM_FORMAT_FLOAT_BE,
SND_PCM_FORMAT_FLOAT64_LE,
SND_PCM_FORMAT_FLOAT64_BE,
SND_PCM_FORMAT_IEC958_SUBFRAME_LE,
SND_PCM_FORMAT_IEC958_SUBFRAME_BE,
SND_PCM_FORMAT_MU_LAW,
SND_PCM_FORMAT_A_LAW,
SND_PCM_FORMAT_IMA_ADPCM,
SND_PCM_FORMAT_MPEG,
SND_PCM_FORMAT_GSM,
SND_PCM_FORMAT_SPECIAL,
SND_PCM_FORMAT_S24_3LE,
SND_PCM_FORMAT_S24_3BE,
SND_PCM_FORMAT_U24_3LE,
SND_PCM_FORMAT_U24_3BE,
SND_PCM_FORMAT_S20_3LE,
SND_PCM_FORMAT_S20_3BE,
SND_PCM_FORMAT_U20_3LE,
SND_PCM_FORMAT_U20_3BE,
SND_PCM_FORMAT_S18_3LE,
SND_PCM_FORMAT_S18_3BE,
SND_PCM_FORMAT_U18_3LE,
SND_PCM_FORMAT_U18_3BE,
};
static const unsigned int rates[] = {
5512,
8000,
11025,
16000,
22050,
32000,
44100,
48000,
64000,
88200,
96000,
176400,
192000,
};
int main(int argc, char *argv[])
{
const char *device_name = "hw";
snd_pcm_t *pcm;
snd_pcm_hw_params_t *hw_params;
unsigned int i;
unsigned int min, max;
int any_rate;
int err;
if (argc > 1)
device_name = argv[1];
err = snd_pcm_open(&pcm, device_name, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
if (err < 0) {
fprintf(stderr, "cannot open device '%s': %s\n", device_name, snd_strerror(err));
return 1;
}
snd_pcm_hw_params_alloca(&hw_params);
err = snd_pcm_hw_params_any(pcm, hw_params);
if (err < 0) {
fprintf(stderr, "cannot get hardware parameters: %s\n", snd_strerror(err));
snd_pcm_close(pcm);
return 1;
}
printf("Device: %s (type: %s)\n", device_name, snd_pcm_type_name(snd_pcm_type(pcm)));
printf("Access types:");
for (i = 0; i < ARRAY_SIZE(accesses); ++i) {
if (!snd_pcm_hw_params_test_access(pcm, hw_params, accesses[i]))
printf(" %s", snd_pcm_access_name(accesses[i]));
}
putchar('\n');
printf("Formats:");
for (i = 0; i < ARRAY_SIZE(formats); ++i) {
if (!snd_pcm_hw_params_test_format(pcm, hw_params, formats[i]))
printf(" %s", snd_pcm_format_name(formats[i]));
}
putchar('\n');
err = snd_pcm_hw_params_get_channels_min(hw_params, &min);
if (err < 0) {
fprintf(stderr, "cannot get minimum channels count: %s\n", snd_strerror(err));
snd_pcm_close(pcm);
return 1;
}
err = snd_pcm_hw_params_get_channels_max(hw_params, &max);
if (err < 0) {
fprintf(stderr, "cannot get maximum channels count: %s\n", snd_strerror(err));
snd_pcm_close(pcm);
return 1;
}
printf("Channels:");
for (i = min; i <= max; ++i) {
if (!snd_pcm_hw_params_test_channels(pcm, hw_params, i))
printf(" %u", i);
}
putchar('\n');
err = snd_pcm_hw_params_get_rate_min(hw_params, &min, NULL);
if (err < 0) {
fprintf(stderr, "cannot get minimum rate: %s\n", snd_strerror(err));
snd_pcm_close(pcm);
return 1;
}
err = snd_pcm_hw_params_get_rate_max(hw_params, &max, NULL);
if (err < 0) {
fprintf(stderr, "cannot get maximum rate: %s\n", snd_strerror(err));
snd_pcm_
Re: [Alsa-user] query alsa for supported sample rates and formats?
On Fri, Apr 17, 2009 at 10:14:16AM +0200, Clemens Ladisch wrote: > Matt Garman wrote: > > Is there a way to query alsa to see what sample rates and > > formats the sound hardware natively supports? > > Try the attached program. Works perfectly. That's exactly what I wanted. Thank you! Matt -- Stay on top of everything new and different, both inside and around Java (TM) technology - register by April 22, and save $200 on the JavaOne (SM) conference, June 2-5, 2009, San Francisco. 300 plus technical and hands-on sessions. Register today. Use priority code J9JMT32. http://p.sf.net/sfu/p ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] query alsa for supported sample rates and formats?
On Thu, Apr 16, 2009 at 07:13:33PM +0300, Sergei Steshenko wrote: > On Thu, 16 Apr 2009 10:49:28 -0500 Matt Garman > wrote: > > Is there a way to query alsa to see what sample rates and formats > > the sound hardware natively supports? > > Something like this: > > " > cat /proc/asound/card0/codec#0 > ... I snooped around in /proc/asound on two different machines, and couldn't find that exact file or even a comparable one. -Matt -- Stay on top of everything new and different, both inside and around Java (TM) technology - register by April 22, and save $200 on the JavaOne (SM) conference, June 2-5, 2009, San Francisco. 300 plus technical and hands-on sessions. Register today. Use priority code J9JMT32. http://p.sf.net/sfu/p ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] query alsa for supported sample rates and formats?
Is there a way to query alsa to see what sample rates and formats the sound hardware natively supports? Thanks, Matt -- Stay on top of everything new and different, both inside and around Java (TM) technology - register by April 22, and save $200 on the JavaOne (SM) conference, June 2-5, 2009, San Francisco. 300 plus technical and hands-on sessions. Register today. Use priority code J9JMT32. http://p.sf.net/sfu/p ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] "bit perfect" audio; was: what provides libasound_module_pcm_plughw.so
On Wed, Apr 15, 2009 at 08:32:11PM +0300, Sergei Steshenko wrote: > If you do not change sample rate, but just increase number of > bits, e.g. 16 -> 24, you lose nothing. Makes sense. But is plughw smart enough to only resample when necessary? I guess that's just something I need to consider if I ever buy new hardware: does it natively support the sample rate and format of all the music I intend to play through it. -- Stay on top of everything new and different, both inside and around Java (TM) technology - register by April 22, and save $200 on the JavaOne (SM) conference, June 2-5, 2009, San Francisco. 300 plus technical and hands-on sessions. Register today. Use priority code J9JMT32. http://p.sf.net/sfu/p ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] "bit perfect" audio; was: what provides libasound_module_pcm_plughw.so
On Tue, Apr 14, 2009 at 10:39:55AM +1200, Eliot Blennerhassett wrote:
> without a default alsa.conf, plughw doesn't exist.
> It is composed of the "plug" on top of "hw".
> On my system it is defined in /usr/share/alsa/alsa.conf
>
> you might try
>
> pcm.!default {
> type plug
> slave.pcm {
> type hw
> card 0
> device 0
> subdevice 0
> }
> }
I have a default alsa.conf in the same location as you.
But, your suggestion did work! Thank you.
Now I have another question: my original goal in using just "hw" was
to send "bit perfect" data to the DAC on my soundcard (i.e. no
software-based alteration of the data/feed the soundcard exactly the
raw audio). The data is just CD digital audio, which I believe is
signed 16 bit little endian/44100 Hz. And then I found that my
soundcard doesn't natively support that.
Now that I'm using plughw---which does any necessary format/rate
conversions---am I really any better than using dmix?
In other words, is the only difference between dmix and plughw that
dmix will mix multiple streams? I don't mean to trivialize mixing
multiple streams, but from my perspective, I'm looking for the best
sound quality for only one audio stream.
So, an even more general question, are there any other sound
"purists" out there, trying to directly use "hw" for bit-perfect
audio? What hardware do you have, what has been your experience,
etc?
Thanks,
Matt
--
This SF.net email is sponsored by:
High Quality Requirements in a Collaborative Environment.
Download a free trial of Rational Requirements Composer Now!
http://p.sf.net/sfu/www-ibm-com
___
Alsa-user mailing list
Alsa-user@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] what provides libasound_module_pcm_plughw.so
On Fri, Apr 10, 2009 at 09:32:13PM -0400, Lee Revell wrote: > Hmm, this all should really have worked OOTB. > > Any way you can revert to the stock Ubuntu 8.10 alsa, disable > pulseaudio, and retest? I'm actually in that state now. My initial post was a bit misleading: I downloaded and built the alsa suite (drivers, libs, utils, plugins) from the "vanilla" (i.e. alsa-project.org) sources, but did *not* install them. (I just built in a temporary directory, and looked for the file in question.) -Matt -- This SF.net email is sponsored by: High Quality Requirements in a Collaborative Environment. Download a free trial of Rational Requirements Composer Now! http://p.sf.net/sfu/www-ibm-com ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] what provides libasound_module_pcm_plughw.so
On Fri, Apr 10, 2009 at 09:16:08PM -0400, Lee Revell wrote:
> OK. I run the exact same distro here.
>
> Normally plughw would be built into ALSA. Not sure why it looks
> for that module.
>
> Try removing /etc/asound.conf and creating this .asoundrc in your
> home directory:
>
> pcm.!default {
>type plughw
>card 0
> }
> ctl.!default {
>type hw
>card 0
> }
Same result:
$ aplay 03_harvest.wav
ALSA lib pcm.c:2156:(snd_pcm_open_conf) Cannot open shared library
/usr/lib/alsa-lib/libasound_module_pcm_plughw.so
aplay: main:583: audio open error: No such file or directory
> Also post the output of "which aplay"
$ which aplay
/usr/bin/aplay
Perhaps that config file format isn't correct, e.g. I just can't use
"plughw" as drop-in replacement for "hw".
Check this out:
$ aplay -D hw 03_harvest.wav Playing WAVE '03_harvest.wav' :
Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
aplay: set_params:954: Sample format non available
Ok, that's what I expected (i.e. same as original config file using
only "hw"). But:
$ aplay -D plughw 03_harvest.wav
Playing WAVE '03_harvest.wav' : Signed 16 bit Little Endian, Rate 44100 Hz,
Stereo
So if I tell aplay directly to use plughw, it works. Thus, my
suspicion is that I need to modify my asound.conf somehow.
Any thoughts?
Thanks!
Matt
--
This SF.net email is sponsored by:
High Quality Requirements in a Collaborative Environment.
Download a free trial of Rational Requirements Composer Now!
http://p.sf.net/sfu/www-ibm-com
___
Alsa-user mailing list
Alsa-user@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] what provides libasound_module_pcm_plughw.so
On Fri, Apr 10, 2009 at 08:44:37PM -0400, Lee Revell wrote: > On Fri, Apr 10, 2009 at 2:13 PM, Matt Garman wrote: > > $ aplay 03_harvest.wav > > ALSA lib pcm.c:2156:(snd_pcm_open_conf) Cannot open shared library > > /usr/lib/alsa-lib/libasound_module_pcm_plughw.so > > aplay: main:583: audio open error: No such file or directory > > > > Building alsa-plugins-1.0.19 and alsa-lib-1.0.19 from source did not > > result in the libasound_module_pcm_plughw.so library. Where can I > > obtain this file? > > What distro? Ubuntu 8.10/Intrepid Ibex. 64 bit. As far as I can tell, no package exists that provides this file, see: http://packages.ubuntu.com/search?searchon=contents&keywords=libasound_module_pcm_plughw.so&mode=exactfilename&suite=intrepid&arch=any I tried searching older and newer releases, and even searched debian's packages. Nothing comes up. Even google doesn't find much when I search for "libasound_module_pcm_plughw.so". -Matt -- This SF.net email is sponsored by: High Quality Requirements in a Collaborative Environment. Download a free trial of Rational Requirements Composer Now! http://p.sf.net/sfu/www-ibm-com ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] what provides libasound_module_pcm_plughw.so
Hello,
I have an MAudio Audiophile 2496 PCI soundcard. My goal is have
ALSA do as little as possible, and just pass the audio to the
hardware.
$ aplay -l
List of PLAYBACK Hardware Devices
card 0: M2496 [M Audio Audiophile 24/96], device 0: ICE1712 multi [ICE1712
multi]
Subdevices: 1/1
Subdevice #0: subdevice #0
So I started with this /etc/asound.conf to bypass dmix:
pcm.!default {
type hw
card 0
}
ctl.!default {
type hw
card 0
}
Unfortunately, it looks like I need to use plughw:
$ aplay 03_harvest.wav
Playing WAVE '03_harvest.wav' : Signed 16 bit Little Endian, Rate 44100 Hz,
Stereo
aplay: set_params:954: Sample format non available
So I modified /etc/asound.conf slightly:
pcm.!default {
type plughw
card 0
}
ctl.!default {
type plughw
card 0
}
$ aplay 03_harvest.wav
ALSA lib pcm.c:2156:(snd_pcm_open_conf) Cannot open shared library
/usr/lib/alsa-lib/libasound_module_pcm_plughw.so
aplay: main:583: audio open error: No such file or directory
Building alsa-plugins-1.0.19 and alsa-lib-1.0.19 from source did not
result in the libasound_module_pcm_plughw.so library. Where can I
obtain this file?
Thanks,
Matt
--
This SF.net email is sponsored by:
High Quality Requirements in a Collaborative Environment.
Download a free trial of Rational Requirements Composer Now!
http://p.sf.net/sfu/www-ibm-com
___
Alsa-user mailing list
Alsa-user@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Spdif clock switching?
On Fri, May 02, 2008 at 11:20:28AM +0200, Bart de Boer wrote: > I got it to work! :D I was telling MythTV to send everything to my > digital output directly. This was wrong. I needed to send > everything to my analog output and let the chip do the work. I'm > now playing music at 44.1 kHz and movies at 48 kHz. :) Would you mind elaborating a bit on what you did exactly in Myth? Thanks, Matt - This SF.net email is sponsored by the 2008 JavaOne(SM) Conference Don't miss this year's exciting event. There's still time to save $100. Use priority code J8TL2D2. http://ad.doubleclick.net/clk;198757673;13503038;p?http://java.sun.com/javaone ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] MAudio 2496 left-channel crackling
On Fri, Apr 25, 2008 at 06:00:01PM +0200, Dominique Dumont wrote: > ok. It you have "lucky time when there's no crackling", then it is > probably a bad solder joint on your card. > > Check the biggest components, they tend to be badly soldered. (I > have repaired the display on my Yamaha DSP-A1 just by resoldering > some components). I did that too---took the card out and visually inspected it for anything obvious (leaking or mis-shapen capacitors, burnt-looking solder joints, etc). I didn't see anything though. But that just means nothing obvious is wrong! ;) Could still be a bunk component in there somewhere. No matter---I got the replacemnt request approved from the retailer, so hopefully I'll have a new one in a couple weeks or so. - This SF.net email is sponsored by the 2008 JavaOne(SM) Conference Don't miss this year's exciting event. There's still time to save $100. Use priority code J8TL2D2. http://ad.doubleclick.net/clk;198757673;13503038;p?http://java.sun.com/javaone ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] MAudio 2496 left-channel crackling
On Fri, Apr 25, 2008 at 04:19:43PM +0200, Dominique Dumont wrote: > Matt Garman <[EMAIL PROTECTED]> writes: > > I forgot to mention, and don't know that this makes any > > difference, but the crackling problems seemed to have developed > > within the last couple weeks or so. For the first month to six > > weeks I had and used it, I don't recall any crackling. And even > > still, I get some "lucky" time when there's no crackling. > > Could you switch the left and right channel (with the RCA plugs) > to check if the crackling noise moves from left to right ? > > This way, you can check if the problems comes from the card or > from the amp. Yup, I tried that, and the crackling noise does move from the left to the right. I've tried all new cables, amps and speakers too. It's definitely coming from the card. Thank you, Matt - This SF.net email is sponsored by the 2008 JavaOne(SM) Conference Don't miss this year's exciting event. There's still time to save $100. Use priority code J8TL2D2. http://ad.doubleclick.net/clk;198757673;13503038;p?http://java.sun.com/javaone ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Sound Quality: difference in digital outputs? optical vs coax? discrete vs integrated?
On Thu, Apr 24, 2008 at 06:29:30PM -0700, Mark Knecht wrote: > In my experience there is no comparison between my Benchmark DAC1 > and the D/A in our home theater. In one case the DAC1 is a $1K > converter driven by a spdif cable. In the other case it's a $300 > Sony receiver with probably a $20 D/A. No comparison as to which > sounds more accurate, open, sweet, clear and any other adjective > you want to use. I get far better sound using the DAC1 into analog > audio inputs than I do using the internal D/A that Sony put in the > receiver. If you don't mind my asking: just out of curiosity, what makes up the rest of the Benchmark DAC1 system? Have you ever heard of the "City Pulse 7.2x II" DAC? It came up in my research of DACs yesterday, and was generally compared to the Benchmark DAC1. - This SF.net email is sponsored by the 2008 JavaOne(SM) Conference Don't miss this year's exciting event. There's still time to save $100. Use priority code J8TL2D2. http://ad.doubleclick.net/clk;198757673;13503038;p?http://java.sun.com/javaone ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] MAudio 2496 left-channel crackling (was: Sound Quality: difference in digital outputs? optical vs coax? discrete vs integrated?)
On Fri, Apr 25, 2008 at 06:17:32AM -0700, Mark Knecht wrote: > Any reason to suspect it's gotten damaged? I have an M-Audio 2496 > and I've never had any trouble like you're experiencing. Nope, I bought it new, and only a couple months ago at that. I forgot to mention, and don't know that this makes any difference, but the crackling problems seemed to have developed within the last couple weeks or so. For the first month to six weeks I had and used it, I don't recall any crackling. And even still, I get some "lucky" time when there's no crackling. > I cannot remember. Did you say whether this machine is dual boot? > Any problems with this specific card in Windows? Nope, it's Linux only. Though I do have an extra hard drive laying around, I suppose I could do a quick windows install and see what happens. > It's my guess that the noise is not being caused by your PC > environment unless you are having some sort of problem with either > the power supply or the motherboard. Testing the card in a second > computer would help identify that. Again, I've used one of these > cards for years and never experienced what you are talking about. > I'm sure if it was happening we would hear about this here all the > time. The motherboard is as new as the soundcard; the system is otherwise 100% stable (and I push it pretty hard---four instances of [EMAIL PROTECTED], h264 video encoding, vmware, lots of compiling (running gentoo), etc). The power supply is probably about a year old, but has never given me any problems (and it's a Corsair, which is supposed to be one of the better brands). Anyway, I bought the 2496 off newegg, and just filled out a replacement request. I think I'll wait until I get the replacement card to do any further testing. Thanks again for all your help! Much appreciated, Matt - This SF.net email is sponsored by the 2008 JavaOne(SM) Conference Don't miss this year's exciting event. There's still time to save $100. Use priority code J8TL2D2. http://ad.doubleclick.net/clk;198757673;13503038;p?http://java.sun.com/javaone ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] MAudio 2496 left-channel crackling (was: Sound Quality: difference in digital outputs? optical vs coax? discrete vs integrated?)
On Thu, Apr 24, 2008 at 09:43:33PM -0700, Mark Knecht wrote: > > One, the crackling of my MAudio 2496. I'm just assuming that > > the inside of my case has a lot of RF interference, so digital > > coming out of the PC seems like a good way to get away from > > that. > > Crackling is not caused by RF. That will cause non-linearity in > the analog portion of the D/A which results in distortion of > levels or 60 cycle hum, but hardly more than that. Just to be clear: are you saying that the crackling on my card certainly doesn't have anything to do with electric noise in my case, and/or power irregularities? > 1) Something in the audio chain, and it can be either analog or > digital, is being over driven. Typically this is caused by > different views of the proper strength of signals. It's often > caused by folks thinking most of the time the song sounds better > when it's turned up loud. My son composes in Acid Pro (as do I) > and we're often finding mistakes within his mixing chain where > little bits of distortion creep in. You should get VERY clear > about how much headroom EVERYTHING in your signal chain has. Much > of this hardware and software has almost none and hence you end up > needing to be very careful about how you drive the signal through > the chain. I usually think of the sound of over-driven circuits as "distortion" rather than "crackling". (That comes from being an electric guitar player, where, for example, the distortion from over-driven tubes is desirable.) Either way, if something is being over-driven, wouldn't that imply that certain types of sounds would be more likely to produce the crackling than others? Also, wouldn't it happen on both the right and left channels? In my case, the crackling is on the *left* channel only (unless I swap right and left on one end of the cable). Plus, the crackling is random---it's not tied to any part of the music I'm playing. In fact it sometimes happens during silent parts of the songs. It also occurs regardless of levels, either on my amp, in alsamixer or in envy24control. I spent quite a while adjusting anything that could be adjusted to see if I could stop the crackling. In fact it's more pronounced when levels are low---it does get louder with increasing volume, but the music gets louder than the crackling, so at sufficiently high volumes, I can almost drown out the unwanted noise. > 2) The second most common cause of crackling, most especially I think > with Linux audio users, is mismatching of sampling rates somewhere in > the chain. Sending a 44.1K Signal across a 48K link, or sending it > across a 44.1K link but the two ends being out of sync with each > other. Ensuring that everything is properly clocked is an absolute > requirement of doing digital audio. That seems more likely to me, but how do I test this? Also, why would this only cause crackling in one channel? - This SF.net email is sponsored by the 2008 JavaOne(SM) Conference Don't miss this year's exciting event. There's still time to save $100. Use priority code J8TL2D2. http://ad.doubleclick.net/clk;198757673;13503038;p?http://java.sun.com/javaone ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Sound Quality: difference in digital outputs? optical vs coax? discrete vs integrated?
In short, I'm looking for the best stereo sound I can get from my Linux system---the main use will be listening to my CD collection (ripped as FLAC). A specific question: if using digital output, is there a difference (sound quality-wise) between optical/toslink and coaxial? Say my soundcard can output both via optical and coax, and my amp also supports both inputs: any reason to choose one over the other? Also, if I'm looking at pure digital output, is there a quality difference between sound card models? My understanding is quite naive (and certainly incomplete!) but it seems like if the software (flac123, mpd, whatever) converts the FLAC code to a digital audio signal, the sound card doesn't really have any work to do. In other words, my motherboard (Abit IP35 Pro) has an optical SPDIF output for its Intel HD Audio system. Would this be any better/worse than, say, a Chaintech AV-710 with optical out (which I'm using in another PC), or an MAudio Audiophile 2496 coax SPDIF output? My understanding has always been that "digital is digital", but after doing some reading today, I'm not so sure that's true. In general, in addition to answering the above questions, I'm kind of looking to continue this thread I found, "Best card for bitperfect SPDIF I/O": http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg21140.html Some general comments/open-ended questions: - I have an MAudio Audiophile 2496. I was using its analog (RCA) outputs to an amp, and then on to some speakers. I noticed a crackling sound in one of the speakers---I've literally changed everything after the sound card (cable, amp, speaker wire, speakers) and the crackling persists. So either I have a bunk card or there's too much RF noise in my case. - I switched back to the onboard Intel HDA chip, and am using its analog output. I *thought* I heard a little bit of crackling last night (but I need to do more listening to verify this). Certainly the analog sound from the integrated chip isn't as good as the MAudio (crackling aside). - It is for the above two reasons that I am looking at going the digital route---if my case is internally noisy, shouldn't digital avoid these problems? - In general, are there any Linux "audiophiles" out there (i.e. using your Linux system as your audio source)? How do you get the signal (analog or digital) out of the computer and into your amp? Thanks, Matt - This SF.net email is sponsored by the 2008 JavaOne(SM) Conference Don't miss this year's exciting event. There's still time to save $100. Use priority code J8TL2D2. http://ad.doubleclick.net/clk;198757673;13503038;p?http://java.sun.com/javaone ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
