[SOLVED] Setting default audio capture device

I have a USB webcam with a microphone that is recognized by ALSA after the on-board line-in. This is fine in Skype because I can set the capture device to the USB microphone, but Flash seems to only use the first capture device (listing only "Linux Microphone") recognized by ALSA. Is there any way to set the default (presumably, the first) audio capture device, or get Flash to see my USB microhone?
Last edited by matkam (2010-09-10 22:01:53)

Solved by creating ~/.asoundrc file containing the following:
pcm.usb
type hw
card U0x46d0x81d
pcm.!default
type asym
playback.pcm
type plug
slave.pcm "dmix"
capture.pcm
type plug
slave.pcm "usb"

Similar Messages

[SOLVED] Setting default audio device pulseaudio

Now i finally get almost all sounds working, but i need to select somehow default output sound device, because i can't change it manually with 'pavucontrol'. When i run Windows applications with Wine i don't get sounds either and 'pavucontrol' is not giving any output of Windows game. Is these problem solved with changing default device if yes, how do i do it?
Using Logitech G930 sound card.
lsof /dev/snd/* output:
COMMAND PID USER FD TYPE DEVICE SIZE/OFF NODE NAME
knotify4 3078 arska134 mem CHR 116,3 1568 /dev/snd/pcmC0D0p
knotify4 3078 arska134 20u CHR 116,33 0t0 3463 /dev/snd/timer
knotify4 3078 arska134 21r CHR 116,3 0t0 1568 /dev/snd/pcmC0D0p
pulseaudi 3218 arska134 mem CHR 116,7 6421 /dev/snd/pcmC1D0p
pulseaudi 3218 arska134 mem CHR 116,8 6422 /dev/snd/pcmC1D0c
pulseaudi 3218 arska134 20u CHR 116,9 0t0 6423 /dev/snd/controlC1
pulseaudi 3218 arska134 26u CHR 116,9 0t0 6423 /dev/snd/controlC1
pulseaudi 3218 arska134 27u CHR 116,8 0t0 6422 /dev/snd/pcmC1D0c
pulseaudi 3218 arska134 34u CHR 116,7 0t0 6421 /dev/snd/pcmC1D0p
pacmd list-sinks output:
Welcome to PulseAudio! Use "help" for usage information.
>>> 1 sink(s) available.
* index: 0
name: <alsa_output.usb-Logitech_Logitech_G930_Headset-00-Headset.analog-stereo>
driver: <module-alsa-card.c>
flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
state: RUNNING
suspend cause:
priority: 9049
volume: 0: 35% 1: 35%
0: -27.42 dB 1: -27.42 dB
balance 0.00
base volume: 100%
0.00 dB
volume steps: 65537
muted: no
current latency: 30.13 ms
max request: 3 KiB
max rewind: 344 KiB
monitor source: 0
sample spec: s16le 2ch 44100Hz
channel map: front-left,front-right
Stereo
used by: 1
linked by: 1
configured latency: 20.00 ms; range is 0.50 .. 2000.00 ms
card: 0 <alsa_card.usb-Logitech_Logitech_G930_Headset-00-Headset>
module: 4
properties:
alsa.resolution_bits = "16"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "USB Audio"
alsa.id = "USB Audio"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "0"
alsa.card = "1"
alsa.card_name = "Logitech G930 Headset"
alsa.long_card_name = "Logitech Logitech G930 Headset at usb-0000:00:13.1-2, full speed"
alsa.driver_name = "snd_usb_audio"
device.bus_path = "pci-0000:00:13.1-usb-0:2:1.0"
sysfs.path = "/devices/pci0000:00/0000:00:13.1/usb6/6-2/6-2:1.0/sound/card1"
udev.id = "usb-Logitech_Logitech_G930_Headset-00-Headset"
device.bus = "usb"
device.vendor.id = "046d"
device.vendor.name = "Logitech, Inc."
device.product.id = "0a1f"
device.product.name = "Logitech G930 Headset"
device.serial = "Logitech_Logitech_G930_Headset"
device.form_factor = "headset"
device.string = "front:1"
device.buffering.buffer_size = "352800"
device.buffering.fragment_size = "176400"
device.access_mode = "mmap+timer"
device.profile.name = "analog-stereo"
device.profile.description = "Analog Stereo"
device.description = "Logitech G930 Headset Analog Stereo"
alsa.mixer_name = "USB Mixer"
alsa.components = "USB046d:0a1f"
module-udev-detect.discovered = "1"
device.icon_name = "audio-headset-usb"
device.intended_roles = "phone"
Solution:
gulafaran wrote:get the wine-pulse package from AUR and add "mmdevapi" to winecfg libraries and set it to disabled and set Audio to Pulseaudio
Last edited by Arska134 (2011-05-15 16:52:53)

and here is some console log when i tried to play Enemy-territory. Sounds not working.
http://paste.servut.us/plain/uyit
I found this:
------- sound initialization -------
SDL audio driver initializing...
SDL_AudioDriverName() = NULL
Why NULL?
Edit:
Sounds not working currently: System sounds, Enemy-territory, Call of duty 4 (with wine)
but works in Wings 2. O_o
Edit 2:
litemotiv wrote:Arska14, please update your topic title so that it reflects that it's a pulseaudio / wine question.
i removed Wine from topic title because it's not affecting wine only.
Last edited by Arska134 (2011-05-15 11:52:25)

[SOLVED]How to set default audio playback device?

I have an X-fi forte soundcard that doesn't work on linux, so I'm just gonna use the onboard realtek chip, but I get no sound from that either unless I physically remove the X-fi card.
I use the X-fi when I boot into windows though, so I'd like to keep it...
How do I change the default playback device?
I didn't find anything that worked in the wiki or with google.
Last edited by raginaot (2010-12-05 04:30:21)

oh wait... I just spotted this in the wiki guide, a little too late
NB: Some modules do not support indexing options eg. snd-hda-intel. If you experience problems on reboot edit /etc/modprobe.d/modprobe.conf to just have the modules.
what does that mean "edit /etc/modprobe.d/modprobe.conf to just have the modules.".... what modules? :S
should I just remove "index=0" from the line?
Edit: yes I should. It's working
Thx man
Last edited by raginaot (2010-12-05 04:29:52)

Media Encoder not finding "audio capture device"

Hi,
Im using Adobe Flash Media Encoder with Dxtory for Twitch streaming, and at first it worked fine. Recorded both my mic and game audio perfectly. Then seemingly suddenly, I fired it up and it wouldn't stream the game audio. It would my Mic, but thats it. I then unplugged the mic and it started saying I had no Audio Capture Device. I checked in Dxtory and its set up the same as it was when it was working. It picks up both the mic and speakers like it should. I have uninstalled this program multiple times, both 3.1 and 3.2 versions, and each time when I re-download it, it loads up the same profile I had before. So when I am uninstalling it, its not uninstalling everything for some reason. Even though the program is no where to be found after its uninstalled. I have downloaded xsplit, and it picks up the game audio and my mic. So its just this program. Anyone have any ideas? I cant seem to find anything online about this issue.
Thanks!

From where? PP sequence? PP bin? PP Source Monitor? Watch folder?

Is there a way to have Firefox respect the default audio output device when it has been changed while Firefox is open?

I have been a Firefox user for many years and I still use it as my primary browser. One problem that has bothered me for years is the fact that when Firefox is open and I change my default audio output device, Firefox doesn't respect that change and continues playing audio through the device that was originally being used. Even if you stop the audio and continue playing, the output device doesn't change.
Is there any way this can be configured in Firefox? Now if I listen to music online (e.g. Soundcloud) I always use Chrome since it's respects the default audio output device change even while in the middle of playing audio.
Reproduce:
1. Start playing audio (e.g. from a YouTube video; HTML5 and Flash are affected)
2. Change your default audio output device
3. Audio will continue to play from original output device
Thanks!

This is the same issue that has plagued me for years. I always thought that this was a Flash plugin issue, but it does occur on HTML5 as well. A friend of mine verified that this does not occur in Linux, but only on Windows. It would be great to have this fixed, as it's honestly the only thing that's making me use Chrome all the time, as I don't want to have multiple browsers open.

No audio capture device detected Osprey 240e

I have recently installed Adobe Flash Media Encoder (AFME). I have installed the Osprey 240e video/audio capture card. At first, the Osprey 240e was detected by the AFME. Now AFME is telling me there is no audio device detected. The video part seems to be fine.
I have:
1. Rebooted the computer AFME is installed on
2. Reinstalled AFME
3. Reinstalled the Osprey 240e drivers (reboot computer again).
When I start the AFME program, it still indicates not audio capture device detected. Any ideas on how to resover the issue.

You may need to open a support case with Viewcast. They have always been very responsive when I have had problems/questions.

How do you select two audio capture devices?

Hi all,
At the moment I'm simply using CaptureDeviceManager.getDeviceList to get a list of all the capture devices on my system. Now when I print this it looks like this...
[DirectSoundCapture : dsound://
LINEAR, 48000.0 Hz, 16-bit, Stereo, LittleEndian, Signed
LINEAR, 48000.0 Hz, 16-bit, Mono, LittleEndian, Signed
LINEAR, 48000.0 Hz, 8-bit, Stereo, Unsigned
LINEAR, 48000.0 Hz, 8-bit, Mono, Unsigned
LINEAR, 44100.0 Hz, 16-bit, Stereo, LittleEndian, Signed
LINEAR, 44100.0 Hz, 16-bit, Mono, LittleEndian, Signed
LINEAR, 44100.0 Hz, 8-bit, Stereo, Unsigned
LINEAR, 44100.0 Hz, 8-bit, Mono, Unsigned
LINEAR, 32000.0 Hz, 16-bit, Stereo, LittleEndian, Signed
LINEAR, 32000.0 Hz, 16-bit, Mono, LittleEndian, Signed
LINEAR, 32000.0 Hz, 8-bit, Stereo, Unsigned
LINEAR, 32000.0 Hz, 8-bit, Mono, Unsigned
LINEAR, 22050.0 Hz, 16-bit, Stereo, LittleEndian, Signed
LINEAR, 22050.0 Hz, 16-bit, Mono, LittleEndian, Signed
LINEAR, 22050.0 Hz, 8-bit, Stereo, Unsigned
LINEAR, 22050.0 Hz, 8-bit, Mono, Unsigned
LINEAR, 16000.0 Hz, 16-bit, Stereo, LittleEndian, Signed
LINEAR, 16000.0 Hz, 16-bit, Mono, LittleEndian, Signed
LINEAR, 16000.0 Hz, 8-bit, Stereo, Unsigned
LINEAR, 16000.0 Hz, 8-bit, Mono, Unsigned
LINEAR, 11025.0 Hz, 16-bit, Stereo, LittleEndian, Signed
LINEAR, 11025.0 Hz, 16-bit, Mono, LittleEndian, Signed
LINEAR, 11025.0 Hz, 8-bit, Stereo, Unsigned
LINEAR, 11025.0 Hz, 8-bit, Mono, Unsigned
LINEAR, 8000.0 Hz, 16-bit, Stereo, LittleEndian, Signed
LINEAR, 8000.0 Hz, 16-bit, Mono, LittleEndian, Signed
LINEAR, 8000.0 Hz, 8-bit, Stereo, Unsigned
LINEAR, 8000.0 Hz, 8-bit, Mono, Unsigned
, JavaSound audio capture : javasound://44100
LINEAR, 44100.0 Hz, 16-bit, Stereo, LittleEndian, Signed
LINEAR, 44100.0 Hz, 16-bit, Mono, LittleEndian, Signed
LINEAR, 22050.0 Hz, 16-bit, Stereo, LittleEndian, Signed
LINEAR, 22050.0 Hz, 16-bit, Mono, LittleEndian, Signed
LINEAR, 11025.0 Hz, 16-bit, Stereo, LittleEndian, Signed
LINEAR, 11025.0 Hz, 16-bit, Mono, LittleEndian, Signed
LINEAR, 8000.0 Hz, 16-bit, Stereo, LittleEndian, Signed
LINEAR, 8000.0 Hz, 16-bit, Mono, LittleEndian, Signed
Now my system actually has 2 michrophones attatched to it. At the moment I'm using DirectSoundCapture to record audio. My first question is, how do you detect multiple audio capture devices? and then of course, how can you record off of different audio capture devices?
Anything would be helpful, if theres a tutorial somewhere on this which i've missed somehow, please tell me. I'm probably missing something pretty major I figure.

I posted in a different thread that I was using the old javax.sound.sampled library to try to record. However, using this library it was quite simple to get a list of devices which could then be used to make a SourceDataLine that could be used to record. However, I was having trouble actually recording and switched to JMF as it has more features and I thought it would be simpler to use. It probably is but I have been struggling. I can record audio in JMF fine but I cant get a device list as I could with the javax.sound.sampled library.
I'll put the code I was working off in here anyway. It shows what I want to do, I just want to do it in JMF.
Note that this code was written 4 years ago by a Matthias Pfisterer and not by myself.
import javax.sound.sampled.AudioFileFormat;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.DataLine;
import javax.sound.sampled.Line;
import javax.sound.sampled.LineUnavailableException;
import javax.sound.sampled.Mixer;
import javax.sound.sampled.SourceDataLine;
import javax.sound.sampled.TargetDataLine;
     /** List Mixers.
          Only Mixers that support either TargetDataLines or SourceDataLines
          are listed, depending on the value of bPlayback.
     public static void listMixersAndExit(boolean bPlayback)
          out("Available Mixers:");
          Mixer.Info[]     aInfos = AudioSystem.getMixerInfo();
          for (int i = 0; i < aInfos.length; i++)
               Mixer mixer = AudioSystem.getMixer(aInfos);
               Line.Info lineInfo = new Line.Info(bPlayback ?
                                        SourceDataLine.class :
                                        TargetDataLine.class);
               if (mixer.isLineSupported(lineInfo))
                    out(aInfos[i].getName());
          if (aInfos.length == 0)
               out("[No mixers available]");
          System.exit(0);
Calling this method on my system will print the following.
Available Mixers:
Microsoft Sound Mapper
AK5370
SoundMAX Digital Audio
AK5370 (2)
This looks very different to what I get when I call CaptureDeviceManager.getDeviceList()
How can I achieve the equivilent in JMF?

Set default audio device?

Prior to installing OSX Yosemite, the system audio output would be automatically switched to my USB audio device as soon as it was plugged in and detected by the OS. In Yosemite, I have to manually switch the output from Internal Speakers to the USB device.
Is there a way to force the setting to change on its own, as it did before?

Hello
I do not want to hard wire the Ipod HIFI. I understand that if I used the line out feature the system would recognize the device. I would like the system to recognize a remote speaker connected by wireless network and to accept this remote speaker as the default audio device. Not likely but should be an option.
Cheers
Warrene
iMac intel Core Duo Mac OS X (10.4.6) Airport Extreme and Express, iPod HIFI, iPods
iMac intel Core Duo Mac OS X (10.4.6) Airport Extreme and Express, iPod HIFI, iPods

"Audio Capture Device not selected" message? Help please!

Howdy folks,
I just got a new PC, Windows 7 OS, I've installed ManyCam and version 3.2 of FME and am looking to broadcast again. I don't recall this issue with my old PC (XP OS, Broadcaster StudioPro virtual cam and FME 2.5). I updated to 3.2 as suggested/recommended in previous posts, but I am getting a message telling me to "please select audio capture devise connected to your system." What am I doing wrong or might there be something that I need to download that I missed?
Any help you can give is greatly appreciated. I truly loved using FME 2.5 but am doing like you suggest here.
Peace,

Hello -
My problem was fixed by setting a recording devise in my "sound" menu.
What I did was -
1. Open the sound menu from the control panel.
2. clicked on the recording tab.
3. Chose "Stereo Mix - Realtek High Definition Audio" as my default device.
That solved it. It was almost too simple. Hope this helps you!

How to change audio capture devices sample rate

I am going to develop a online voice chat applet. Now everything works fine. But I didn't find anyway to capture audio in a lower sample rate. The default sample rate is 441000Hz. I am using MediaLocator to capture audio and pass it into Processor.
MediaLocator ml = new MediaLocator("dsound://");
DataSource ds;
DataSource clone;
try {
ds = javax.media.Manager.createDataSource(locator);
} catch (Exception e) {
return "Couldn't create DataSource"+e.getMessage();

Use GSM format and do it with setting of BitRateControl. It is better than convert the sample rate of your capturing device. I suppose that you try to reduce load of network traffic with changing sample rate, right?
Best regards from Germany,
r.v.

Wasapi: IAudioClient- Initialize() hanging when opening audio capture device

I'm trying to open the capture/input audio device. I call ActivateAudioInterfaceAsync() , then when I call IAudioClient->Initialize() on the capture device, the app hangs, supposedly because for some reason it's trying to display the 'Allow app to
use the mic?' prompt.
The hang _doesn't_ occur if I manually set the app permission to use the mic (i.e. launch the app without having it open the audio devices, open the permission tile, activate the mic permission).
Calls to ActivateAudioInterfaceAsync and IAudioClient->Initialize() are made from the UI thread (the one that receives callbacks for buttons, which on Metro style apps seems to be different from the thread that starts the application, i.e. WinMain).
What am I doing wrong?
I've attached the stack that shows the app hung below.
Thanks,
Flavio.
ntdll.dll!_NtAlpcSendWaitReceivePort@32()
Unknown
rpcrt4.dll!LRPC_CASSOCIATION::AlpcSendWaitReceivePort(unsigned long,struct _PORT_MESSAGE *,struct _ALPC_MESSAGE_ATTRIBUTES *,struct _PORT_MESSAGE *,unsigned long *,struct _ALPC_MESSAGE_ATTRIBUTES *,union _LARGE_INTEGER *)
Unknown
rpcrt4.dll!LRPC_BASE_CCALL::DoSendReceive(void)
Unknown
rpcrt4.dll!LRPC_BASE_CCALL::SendReceive(struct _RPC_MESSAGE *)
Unknown
rpcrt4.dll!LRPC_CCALL::SendReceive(struct _RPC_MESSAGE *)
Unknown
rpcrt4.dll!_I_RpcSendReceive@4()
Unknown
rpcrt4.dll!_NdrSendReceive@8()
Unknown
rpcrt4.dll!@NdrpSendReceive@4()
Unknown
rpcrt4.dll!_NdrClientCall2()
Unknown
AudioSes.dll!_AudioServerCreateStream@36()
Unknown
AudioSes.dll!CAudioClient::CreateRemoteStream(enum SYSTEM_AUDIO_STREAM_TYPE,__int64,__int64,struct SYSTEM_AUDIO_STREAM *)
Unknown
AudioSes.dll!CAudioClient::Initialize(enum _AUDCLNT_SHAREMODE,unsigned long,__int64,__int64,struct tWAVEFORMATEX const *,struct _GUID const *)
Unknown
Metro.exe!App1::MainPage::Startup() Line 171
C++
Metro.exe!<lambda_f0ad93afb1643234338c56b595dca446>::operator()() Line 48
C++
Metro.exe!Platform::Details::__abi_FunctorCapture0<<lambda_f0ad93afb1643234338c56b595dca446>,void>::Invoke() Line 850
C++
Metro.exe!Windows::UI::Core::DispatchedHandler::Invoke()
C++
Metro.exe!Windows::UI::Core::DispatchedHandler::[Windows::UI::Core::DispatchedHandler::__abi_IDelegate]::__abi_Windows_UI_Core_DispatchedHandler___abi_IDelegate____abi_Invoke()
C++
Windows.UI.dll!6283fdff()
Unknown
[Frames below may be incorrect and/or missing, no symbols loaded for Windows.UI.dll]
Windows.UI.dll!628314ae()
Unknown
user32.dll!_InternalCallWinProc@20()
Unknown
user32.dll!_UserCallWinProcCheckWow@36()
Unknown
user32.dll!_CallWindowProcAorW@24()
Unknown
user32.dll!_CallWindowProcW@20()
Unknown
Windows.UI.Xaml.dll!5e06b434()
Unknown
Windows.UI.Xaml.dll!5e06b3af()
Unknown
user32.dll!_InternalCallWinProc@20()
Unknown
user32.dll!_UserCallWinProcCheckWow@36()
Unknown
user32.dll!_DispatchMessageWorker@8()
Unknown
user32.dll!_DispatchMessageW@4()
Unknown
Windows.UI.dll!628311dc()
Unknown
Windows.UI.dll!62831290()
Unknown
Windows.UI.Xaml.dll!5e16dee2()
Unknown
Windows.UI.Xaml.dll!5e16dea1()
Unknown
Windows.UI.Xaml.dll!5e16de65()
Unknown
ntdll.dll!_RtlReleaseSRWLockExclusive@4()
Unknown
03655668()
Unknown
twinapi.dll!67b2c9dd()
Unknown
twinapi.dll!67b2cab2()
Unknown
twinapi.dll!67b2c9f6()
Unknown
SHCore.dll!740c1ffd()
Unknown
kernel32.dll!@BaseThreadInitThunk@12()
Unknown
ntdll.dll!___RtlUserThreadStart@8()
Unknown
ntdll.dll!__RtlUserThreadStart@8()
Unknown

When you're in this state, can you take a dump of the svchost.exe process containing the AudioSrv service and send it to me? email to (mateer at microsoft dot com)
Matthew van Eerde

Set default audio track for playback in Source monitor

So I'm an editor that is somewhat new to the profession and I am working on a project right now with close to 100 video clips in my project. Most of the clips have their audio recorded to track 2 but when I load the clips into the source window to find the footage asked for in my script there is no audio playback.
They play fine in the program monitor once placed in a timeline, because I have all the audio tracks are turned on until I do my mastering at the end of the project. To get audio playback in the source monitor I have to change the view option to audio waveform which shows a blank screen because there is no audio on track. I then go to the drop-down menu just below the video display and select track 2 at which point the audio waveform shows up and plays back fine. This isn't too arduous a process and isn't a great burden but it slows my workflow down considerably as I have to repeat the process for each new clip I load into the source monitor.
My question then is if there is anywhere in Premiere that I can set which audio track is the default audio playback track for clips loaded in the source monitor. I work on a lot of similar projects like this for the same client and they don't have the most rigorous standards for their shooters. This problem has come up before and will most assuredly arise again in the future so any tips/advice on how to fix this on my end would be much appreciated.
Thanks!

I'm not really sure.I don't do any shooting for the client and in fact don't have any contact with the shooters, we just do the post. They use a number of different freelance videographers all of whom have different equipment and procedures when it comes to formatting, orginization, etc.

How to set preferred audio output device on macbook pro mavericks

I've got a pair of Harman Kardon bluetooth soundsticks that I use as my living room speakers. Some time ago, I connected my macbook pro to them via bluetooth, and now whenever I restart my macbook or move it away from the apartment and then move it back, it automatically finds the soundsticks, connects to it via bluetooth, and sets it as the default sound output device.
I would like the soundsticks to remain connected whenever they're within range, but I want my macbook's internal speakers to be the default sound output even if the soundsticks are connected. I can't seem to set output device preferences in the sound preferences panel.
thanks,
- will

I dont have a solution, but there is a thread on this topic already. https://discussions.apple.com/message/23679152#23679152

Need help getting current/default audio playback device from VISTA

Hello,
I need to get the current audio playback device in VISTA.
It's easy enough in XP as I just get this information from the
registry. My understanding is that things are different under
VISTA. I was told you can't change the device but you can get the
device using the following...
--CoCreateInstance(IMMDeviceEnumerator)
--IMMDeviceEnumerator->GetDefaultAudioEndpoint(eConsole)
Problem is I don't know what to do with that. I assume it
needs to be compiled with some library (compiler, library, etc... I
don't have)
So... question, is there already an xtra that can get this
information OR... can I pay someone to make an xtra using the above
info?
If you are able/willing to help, please let me know asap as
project is due next week. I can pay you through paypal.
Thank you!!!!!
[email protected]

http://www.schmittmachine.com/asPitchDetect.html
Antoine Schmitt at that url builds Xtras and his
asPitchDetect Xtra can get info on recording devices... try
contacting him and he might be able to help you with a custom Xtra
to get info on playback devices in Vista.

How to set default audio device for java apps on Linux?

I would like to find a way to choose a particular (ALSA) audio device for java applications. How can this be done without reprogramming the application or changing the default ALSA device? I came across "sound.properties", but could not find any information how to use this file on an ALSA system.
Thanks for any help, Mark

Eugenia wrote:Well, it's still not good enough.
We can't satisfy everyone. What you want is a corner case. For most people ALSA and dmix works appropriately, and that is what esd uses by default. We cannot document and automate every configuration possible. That's up to the users.
However, if you don't want to restart gnome, try the below... I make a few assumptions, but even if they're wrong, they're hardly difficult to work around.
Just copy the /etc/esd.conf to ~/.esd.conf, make your change, then do killall esd. Presumably gnome will restart esd, and it ought to read your local config. If it doesnt, start it yourself.
Feel free to write a script that automates this, or document it on the wiki.
I have to restart Gnome to get this thing working. Other OSes *automatically* can use an alternative device. But if I change ESD, I will have to restart Gnome, no?
Well, if you try my suggestion above, no. You can restart esd yourself. And that's what all the other distros do, except with some fancy GUI.
James

[SOLVED] Setting default audio capture device

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