Sample .asoundrc files

A plugin or plug-in is a computer program that can, or must, interact with another program to provide a certain, usually very specific, function. Typical examples are plugins to display specific graphic formats e. The main program a web browser or an email client, for example provides a way for plugins to register themselves with the program, and a protocol by which data is exchanged with plugins.

What does it do, what does it mean, what for do I need it? The slave is the device that is controlled by the plugin, and recieves the plugin audio output in the case of playback, or provides input for recording. This defines a slave without any parameters.

It's nothing more than another alias for your sound device. The slightly more complicated thing to understand is that parameters for 'pcm types' must be defined in the slave-definition-block.

Let's setup a rate-converter which shows this behaviour. This automatically converts your samples to a It's not very useful because most players, and ALSA, convert samples to the correct sample-rate that is supported by your sound card but you can use it for a conversion to a lower static sample-rate, for example. A more complex tool for sample conversions is the PCM type "plug".

The syntax is:. If you use aplay with the verbose option -v you will see the settings from the original file. For example,. I had a lot of trouble first figuring out how I could split front and rear channels into two devices that could be used independently. The following.

It can be used with 'mplayer', for example, as follows:. Actually this is a bug and has already been fixed in versions higher than 1. If your card has a number of stereo sub-devices that operate synchronously, you can join them into one virtual multichannel device. The following joins two adjacent sub-devices into a 4 channel device. You can use this device with JACK. In this example an intel8x0 ICH6 hw:0 and an Aureon 5.

The default device is a stereo device, the audio stream is duped to both cards. Dmix is enabled on both cards. For a detailed description of the syntax of the. Joern Nettingsmeier posted an. If you are interested in a more advanced. The sample rate is correct and the latency looks good a 5. No xruns in the message window. Looks like that could have been the problem. Thanks, to you and everyone that contributed.

You can try to set the number of input channels higher. It only works for me on an Phase88 also with envy24 chip, when i use 12 channels 8 analog input, 2 spdif, 2 intermal mixer. You might have to try every number between 4 and your number of inputs plus 4 until you find the right one. Then it will complain about the playback channels, because of the same problem. For me playback channels are Try to set them as well and jack should start running.

If i remember right, this module is still available. In the standard kernel of edgy. Maybe jackstart was used or jack in ubuntu was compiled to use this module what i doubt.

You should definitely not specify -i2 -o2 with an ice card, just let jackd work it out by itself as Paul suggested. If you want to see how many channels are running you can use jackd -v option. Low software latency is only required for MIDI etc. You have libjack0. Could that be one of the problems? I seem to remember seeing this message with an earlier version of jackd 0. You should use the largest buffer size possible for reliability which is for ice - remember you have zero latency hardware monitoring.

Changing the sample rate in version. If you start jack from commandline with these options: instead of hw:1 you might need hw:0 or hw:2 or whichever number your soundcard has jackd -d alsa -d hw:1 -r Then jack returns something like that: jackd 0.

Here is the output: jackd 0. It seems there is no problem with jackd setting the sample rate to I'm currently using this USB mic and the 3. I am using the same setup. I do however hear "front left, front left And now trying to run python -m googlesamples. How do you know the recording actually picked up sound? Is the Connect Failed error persistent? Did it work for you before? Are you running any kind of local or network firewall? It's supposed to record something for 5 seconds and then replay it?

Does it behave correctly? After reading the comments I changed the. Now the mic and playback are working fine. However on starting the SDK the same error pops up.

Can you paste the output of python -m googlesamples. How do I change the sample rate of the audio devices. I did not, however I haven't had much time to test again And here's what I get when I try to run python -m googlesamples. Do I need the full version for this to work? I just assumed that it couldn't be the USB mic that was the problem That happens when the connection timeout and we automatically retry the connection with 7da41cf.