I don't get sound from usb-audio, making sure the name matches.
Could the .asoundrc syntax be wrong there?
I doubt the following could be wrong:
pcm.usb-audio {
type hw
card 0
}
Unless usb-audio is not your device. I would tend to take mpd out of the
pcm.my_device
Sorry, thought it would be understood that 'my_device' is the alsa alias
(in my case the the name of the kernel module without the
leading snd-) for the hardware device in question (see your
modules.conf file or whatever is proper for your distro). In my case I
have 2
On Friday 13 June 2008, Grant wrote:
I changed my config like so and restarted alsasound with the same
results:
.asoundrc:
pcm.usb-audio {
type hw
card 0
}
pcm.usb-audio_44 {
type plug
slave {
pcm usb-audio
rate
I changed my config like so and restarted alsasound with the same
results:
.asoundrc:
pcm.usb-audio {
type hw
card 0
}
pcm.usb-audio_44 {
type plug
slave {
pcm usb-audio
rate 44100
}
}
On Friday 13 June 2008, Grant wrote:
I don't get sound from usb-audio, making sure the name matches.
Could the .asoundrc syntax be wrong there?
I doubt the following could be wrong:
pcm.usb-audio {
type hw
card 0
}
Unless usb-audio is not your device. I would tend
of the most obtuse syntax in Linux.
Demian Martin
Product Design Services
-Original Message-
From: Rene Herman [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 11, 2008 10:29 PM
To: Demian Martin
Cc: alsa-user@lists.sourceforge.net
Subject: Re: [Alsa-user] Output sample rate
On 12-06-08 07
On 12-06-08 08:28, Demian Martin wrote:
I don't think we are in disagreement in substance. I was trying to
give a larger framework for all of those not as familiar with the
general workings and decisions behind computer audio as currently
implemented. Just works is very important to most
dmix *is* an application, conceptually it is not different to esd, artsd
or pulseaudio opening the ALSA hardware directly. It just happens to be
an ALSA plugin and is a part of the signal chain that ALSA-lib sets up
by default.
You can either:
a) configure dmix using a custom .asoundrc to
On Thursday 12 June 2008, Grant wrote:
I'm trying to get music from mpd to my USB DAC in 100% untouched
form. My mpd.conf is as follows:
audio_output {
type alsa
name USB Monica
device hw:0,0
format 44100:16:2
}
Don't know anything about mpd but you can set up an asoundrc plug in
alsa
I'm trying to get music from mpd to my USB DAC in 100% untouched
form. My mpd.conf is as follows:
audio_output {
type alsa
name USB Monica
device hw:0,0
format 44100:16:2
}
Don't know anything about mpd but you can set up an asoundrc plug in
alsa that will fix the rate from alsa to the
On Friday 13 June 2008, Grant wrote:
pcm.my_device
Sorry, thought it would be understood that 'my_device' is the alsa alias
(in my case the the name of the kernel module without the
leading snd-) for the hardware device in question (see your
modules.conf file or whatever is proper for your
Where is my output sample rate defined? I'm trying to make sure mpd
isn't resampling my music before it's sent to the USB DAC.
- Grant
-
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Grant wrote:
Where is my output sample rate defined? I'm trying to make sure mpd
isn't resampling my music before it's sent to the USB DAC.
If you are playing audio that is in recognized format the rate is
defined within the audio file itself and was set at time of creation.
If you want
-Original Message-
From: Grant [EMAIL PROTECTED]
To: alsa-user@lists.sourceforge.net
Date: Wed, 11 Jun 2008 15:35:42 -0700
Subject: [Alsa-user] Output sample rate
Where is my output sample rate defined? I'm trying to make sure mpd
isn't resampling my music before it's sent
On 12-06-08 00:35, Grant wrote:
Where is my output sample rate defined? I'm trying to make sure mpd
isn't resampling my music before it's sent to the USB DAC.
Check /proc/asound/card0/pcm0p/sub0/hw_params while mpd is playing (for
a suitable value of (0,0,0) ofcourse).
Rene.
On 12-06-08 02:17, Sergei Steshenko wrote:
From: Grant [EMAIL PROTECTED]
Where is my output sample rate defined? I'm trying to make sure mpd
isn't resampling my music before it's sent to the USB DAC.
I initiated a similar thread recently.
The short answer - nowhere.
As I was
-Original Message-
From: Rene Herman [EMAIL PROTECTED]
To: Sergei Steshenko [EMAIL PROTECTED]
Date: Thu, 12 Jun 2008 04:12:58 +0200
Subject: Re: [Alsa-user] Output sample rate
The sampling rate is a property inherent to the data.
Rene.
???
Are trying to tell me that sample rate
On 12-06-08 05:52, Sergei Steshenko wrote:
From: Rene Herman [EMAIL PROTECTED]
The sampling rate is a property inherent to the data.
Are trying to tell me that sample rate is inherent to analog source connected
to microphone or line input ?
Oh, not again... please get a clue. DATA.
Rene.
-Original Message-
From: Rene Herman [EMAIL PROTECTED]
To: Sergei Steshenko [EMAIL PROTECTED]
Date: Thu, 12 Jun 2008 05:55:40 +0200
Subject: Re: [Alsa-user] Output sample rate
On 12-06-08 05:52, Sergei Steshenko wrote:
From: Rene Herman [EMAIL PROTECTED]
The sampling rate
On 12-06-08 06:30, Sergei Steshenko wrote:
Yes again - to me ALSA's sample rate implementation looks quite
illogical - IMO it should be the other way round - user first
mandates sample rate, and then playback sources adapt through
resampling if necessary.
Great setup once we have infinitely
Its very simple.
Most sound devices support a number of sample rates. Common ones include
16 bit @ 44.1khz, 16-bit @ 48khz, 24-bit @ 96 khz etc.
Only one application has exclusive control over the sound hardware at
any time.
Whatever rate that application opens the soundcard at, is the rate
On 12-06-08 06:53, Pete Black wrote:
Its very simple.
Most sound devices support a number of sample rates. Common ones
include 16 bit @ 44.1khz, 16-bit @ 48khz, 24-bit @ 96 khz etc.
Only one application has exclusive control over the sound hardware at
any time.
Whatever rate that
Computer audio and sample rate issues are popping up everywhere, driven by
the desire for high quality audio on PC's finally. On Windows and Mac's its
even harder to get it right.
In Alsa (and PC audio architecture in general) the system has a default
sample rate, usually set by the driver it
dmix *is* an application, conceptually it is not different to esd, artsd
or pulseaudio opening the ALSA hardware directly. It just happens to be
an ALSA plugin and is a part of the signal chain that ALSA-lib sets up
by default.
You can either:
a) configure dmix using a custom .asoundrc to
On 12-06-08 07:13, Demian Martin wrote:
Computer audio and sample rate issues are popping up everywhere, driven
by the desire for high quality audio on PC's finally. On Windows and
Mac's its even harder to get it right.
In Alsa (and PC audio architecture in general) the system has a
On 12-06-08 07:17, Pete Black wrote:
dmix *is* an application, conceptually it is not different [ ... ]
Let's call it a conceptlication then. As said, your basic description
was correct.
Rene.
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