What i am trying to acomplish is this
Code:
I have found a few guides that were to accomplish this, but they seem outdated as options they call for are not available to me.
I have a headset microphone plugged into the built-in audio input. The output is plugged into an amplifier > speakers.
Pulseaudio is "helpfully" routing the microphone input signal to the speaker output. I do not want this. I go to pavucontrol and disable the "monitor of the internal audio analog stereo" -- and what I say into the microphone is still coming out of the speakers.
Google gives a lot of information about how to live-stream mic input over the Internet. And, I found this here (that's for karmic, I'm running lucid):
[URL]
... but that's about how to ENABLE loopback. It says loopback should not be enabled by default, but I very clearly hear that something is doing the same thing as loopback.
(Of course, I need to hear sound from applications that are playing, and when I'm dictating text into NaturallySpeaking running under virtual box, I need sound input to go there -- so turning the mic all the way down to zero is not the solution.)
I'm fairly sure I remember hearing this behavior ever since installing Ubuntu, and I didn't do anything to turn it on. I can't imagine most users would prefer this behavior -- if it's the default behavior, why? It makes absolutely no sense to me. (Sorry. It's one of those piddly little configuration things that costs a couple of hours of web searching time, and after a while I just have to give up, but not before becoming much more frustrated than I should have to.)
So, to be clear, here's what I want:
- Applications playing audio should send sound to the main output.
- Applications receiving audio input should hear the microphone.
- Microphone sound should not EVER go to the main output unless I have explicitly launched an application making the connection. (The signal goes input -> output even when NO application is open - hence my consternation.)
I am currently trying to start doing some screen casting and having a problem with audio recording in gtk-recordmydesktop. The problem is that I need to record both; my computers audio card output, and my mic input at the same time. As of now I am able to record one or the other.
View 4 Replies View RelatedWhen I installed my 64 bit system of Ubuntu 10.04 the sound worked very well and I were very happy. The problem started however when I installed Skype which uses pulseaudio. As soon as I start skype (or any other application that uses pulse, HoN for example) the applications sound output or input does not work at all. If I have pulseaudio started in some way, applications that I suppose do not use it like spotify or flash player stops to produce sounds. And when I type "pulseaudio" in the terminal it gives me this:
[Code]...
I'm getting silence when trying to record output from pulseaudio on my laptop. This wasn't happening the last time I tried, which, I admit, was almost certainly a previous ubuntu release (I'm on 10.10).
My sound card is an Intel ICH8-family device, supported by the snd_hda_intel module. PulseAudio calls it "Internal Audio Analog Stereo".
I'm doing various things that I find on the onlines, a few main ideas:
1. parec
I use:
Code:
parec -d alsa_output.pci-0000_00_1b.0.analog-stereo.monitor
When I search for 'monitor' in $(pactl list), the only thing listed is alsa_output.pci-0000_00_1b.0.analog-stereo.monitor.
I use sox to convert the output to wav:
Code:
sox -t raw -r 44k -e signed-integer -L b 16 -c 2 $RAWFILE $WAVFILE
[Code].....
I've got two soundcards in my new machine - Realtek ALC8892 onboard, with headphones plugged in, and an M-Audio Audiophile 192 with speakers. There's also HDMI sound out on the graphics card, but I don't use that.
Audio with PulseAudio's output through the sound card is fine. Audio with PulseAudio's output through the onboard sound is fine. If I set the output to Simultaneous Output, however, I get significant lag. Almost half a second. It's quite distracting.
Is this just a problem with PulseAudio's simultaneous output implementation, or is there a fix that I've overlooked? Running 64-bit Natty, btw.
sat in an internet cafe with my music on loud using my headphones when my cell phone vibrated and upon taking off my headphones I noticed that pulseaudio was channeling all my music not only to my headphones but also to my regular monitor out!
So I was actually listening to music using my headphones when in fact the sound was LOUD on my laptop speakers...
Is there a way to configure the thing to not do that? I want the sound to be muted as soon as I plug in my headphones.
I don't want to monitor my inputs. In other words: My microphone input plays through my speakers, adding a lot of undesirable noise from the mic. I can mute the mic inputs in alsamixer and/or in the Pulse volume control. This works to get the mic out of the speakers--but it also mutes my mic entirely, so I cannot record audio or use Google Talk. I want the microphone muted in my output, but available to my apps.
(I try to get on the Pulse bandwagon but it is so hard--I thought this was exactly what it was supposed to do, give easy mixing control over inputs and outputs. But I have to search for "fixes" with every single Ubuntu release for two years now, ever since it was first included. Ok, venting over, time to get it working!)
Today I tried to use padevchooser on Natty to send my laptop audio to my home media machine (which has the good speakers) using the "default server" option. Turns out padevchooser doesn't work on Unity and the threads I have seen say it is deprecated for other gui tools,
So what is the easy gui way to switch from my local pulse server to another one on my LAN, without using padevchooser, or switching off Unity? Or is this a regression?
Have you ever wanted to record a sound heard from your computer or website? If yes outRec is perfect for you. This simple but powerful application let you to record your sound card audio output easily in a few steps and save it in different kinds of formats like wav, mp3 or ogg.
Download from [URL]
To get it work you need to install the dependencies with:
Code:
sudo apt-get install sox gambas2-runtime lame mplayer libnotify-bin libmp3lame0 gambas2-gb-form gambas2-gb-desktop gambas2-gb-gui libtwolame0
I am attempting to record my microphone input using ffmpeg / alsa but always receive the following error:
[code]...
Using arecord works perfectly with hw:0,0. I had also tried to use /dev/snd/hwC0D0 as input but I receive an error from alsa saying that it doesn't exist (despite it existing and having adequate permissions). I know absolutely nothing about hardware but am comfortable around the command line
I'm using pulse audio sound server in opnsuse 11.3 and latest version of flash player, The problem is playing video or sound with flash player in firefox, Once I play something in firefox sound output it seem like closed to other applications like amarok.
View 3 Replies View RelatedI have recently acquired a Lenovo Q150 machine and attempting to use it as a HTPC. I've been reading that with this platform a newer kernel is required to make wireless, sound and a few other tweaks work correctly--so I bumped up to testing repositories to upgrade to the 2.6.38 kernel.[URL]...
The audio on this device has been more than a pain. I'm currently using XBMC to play media on this device and after setting the outputs to custom: plughw:1,9 sound is played correctly. I found this out by using alsamixer, selecting the sound card with the F6 key (Nvidia 1) unmutting all outputs, quiting, and running speaker-test -D plughw:1,X where X is the sub-device from the output of aplay -l until sound could be heard from the receiver.
Now my problem is that applications like mplayer, and iceweasel won't output any sound. I'd prefer not to use the optical out on the device and would like to send sound over HDMI. Has anyone had any luck getting it to work as it should?
I've also installed pulseaudio, not too sure if this is really needed. I've also used module assistant before upgrading to compile alsa from source, it worked but i just decided to upgrade the kernel instead of dealing with m-a every time an update comes through. Linux floppy 2.6.38-2-amd64 #1 SMP Thu Apr 7 04:28:07 UTC 2011 x86_64 GNU/Linux
apt-cache policy alsa-base
alsa-base:
Installed: 1.0.23+dfsg-2
Candidate: 1.0.23+dfsg-3
[Code]...
A few days ago I setup a pc. It contains an "Intel ICH5" sound card. Before I begin posting all the configuration stuff I've tried so far what the standard procedures are to set it all up for recording the sound card output (with Audacity)?
On my older computer it all worked fine but now I only get silent recordings. However sometimes (when I played on the controls the right way...) I get recordings where you can (amoung some noise) recognize the stuff I actually wanted to record very low... Apart from Audacity I've also tried
Code:
arecord -vv -fdat audio.wav
which didn't work either. That's why I think the problem is not related to Audacity but to the system.
Is the combination of OpenSuse 11.2 + ALSA + Intel ICH5 commonly known to cause such trouble?
I am attempting to connect a bluetooth headset to a Telex Intercom System. I can connect my bluetooth headset to Ubuntu and use the microphone and the speaker perfectly. There are no problems with the Line-In or the Line-Out.
Is there any way to connect the BT-Microphone to the Line-Out while connecting the Line-In to the BT-Speaker?
I use the growisofs command from the terminal to do most of my burning. Mostly because i burn almost the exact same type of file every time so its pretty easy to set up. I've done this dozens of times and never once had a problem before. I loved it because it was so much simpler than doing this in windows where i would need to install a few extra programs and codecs. That was up until today where it started failing every single time i tried to burn something.
For some reason every time i try to burn a disc it gives me this error message:
Code:
I'm really at a loss as to why this is happening. Every thread i found that discussed this problem was mostly focused on k3b or brasero and never really answered the question. As far as i know nothing has been updated since the last time i burned a disc a few days ago. Everything is exactly the same as it was when it worked. The only thing i could think of that might have caused this sudden failure is that i flashed the firmware of another cd drive using the same sata port that this drive is currently plugged into, but that was in windows so i dont see how it could have affected this drive.
For some reason my soundcard is recording every playback sound. This become apparent when I was trying to make a Skype call and the other people couldn't hear me. I checked my sound preferences and noted that the input was recording what they say (they could hear an echo). Also if I played back anything it would redirect to them. This has never happened to me and I've made a lot of calls in the past.
I tested with Audacity and it records fine, but when I play back the recording, it redirects to the input channel so it's not a problem with Skype. (my soundcard is a VIA VT1708B 8-Ch). Somewhere I read that the output of aplay -l is useful.
Here it is:
I'm using FFADO on JACK in order to use my firewire device (M-Audio Profire 610) and I seem to have the opposite issue as others who have similar problems. Everything starts up fine, and I can record audio into Ardour using my inputs, but I can't get any output from any of my 8 line outs.
I've checked the connections using QjackCtl, and I tried to route audio from another program (Audacious) but I can't seem to pull audio out of the device (or the computer won't send my device audio?).
I used to be able to use Skype without a problem. I must have played with something as it now doesn't allow me to both use a microphone and external speakers at the same time.I go into sound setting and I am able to test the microphone (works) but then the speakers don't show, if I reverse the tests then the opposite happens.
View 1 Replies View RelatedIs it possible to output one playback stream to multiple devices simultaneously with the current PulseAudio / Phonon setup? The PulseAudio mixer only has radio buttons to choose one device per playback stream. I believe the hardware is capable of this, since I remember doing that before we had PulseAudio. How can I duplicate an audio stream?
Here's one application scenario: I am travelling with my family, all crammed in small hotel room. My wife and me want to watch a movie on my laptop without waking up our kids. I just happen to have one analogue headphone available and one wireless USB headset with me. (Of course, the low tech solution is to bring an 3,5mm Y-cable to attach two analogue headsets, but I would really love to use the USB headset together with the analogue one.)
Another similar thing that bugs me is that my laptop's built-in speakers now always seem dead when an analogue headphone is plugged in. This is mostly what one wants, and before PulseAudio, one had to manually switch them off which was generally annoying. However, the downside is for example with notifications.
For example, before PulseAudio, I could configure Skype to always ring over the laptop's built-in speakers, regardless of whether the analogue headphones were plugged in. This is no longer possible, since PulseAudio does not distinguish between built-in speakers and built-in analogue port any more, while old Alsa did. So in my office, where some analogue headphones are plugged into the docking station, I never hear Skype ringing if I don't wear the headphones.
Working on backing up my DVD library. Anyone know why Brasero would fail to copy data due to an input/output error? Movie was Toy Story 2, disc 1.
View 6 Replies View RelatedI'm trying to create an iso file in a terminal with the following command:
$cat /dev/sr0 > nameofdisk.iso
I get the following error
cat: /dev/sr0: Input/output error
I already checked and my optical drive is indeed /dev/sr0. I've hunted google a few hours trying to figure it out. Does anyone know why I'd be getting this error?
I have a TV card connected to my Dish box, and I would like to use it on my computer. The tv card works fine, and the audio is connected to my line in. The audio card receives the audio signal, and I can set the volume and record it without any problem, but I can't get it to redirect to an output in order to hear the sound. I have been battling with this issue since a kernel update back when 11.2 was about to come out. Back then, the alsa mixer had an option called audio loop-back, which did the trick. Since I know I ca record the audio and play it back without any problem, is there an application that can do this automatically and simultaneously?
View 1 Replies View RelatedI can play audio in my alsa system and I can record. To keep things simple, I use arecord to record and aplay to play. So far so good. The problem is, I cannot hear what I am recording as I record it. Is there some way to get the input to be echoed to the output? I would optimally like to be able to do this before recording as well as during, so that I can check the levels, find the part to record on the tape, beforehand.
I have tried a test of my card: I had aplay running in one term and arecord running in another. They both ran fine and when I checked the just-recorded file, it was different audio from what I had been playing. This would indicate that the card is able to input and output at the same time, I think? I've done the usual googling as well as reading through URL... and I have done many passes of trial-and-error on my .asoundrc. The more I look at my .asoundrc, the more it looks like it should be doing what I want, but it is not.
My .asoundrc (which again, does works but does not echo audio input):
Code: Select all# pros: this allows playback and record. Playback of several
# programs at the same time works (their outputs are mixed
# together, mono and stereo mixed to stereo)
#
# cons: input is never echoed!
[code]....
Because I am trying to use skype, I try to record my voice with a microphone.
I am working in FC12.
Code:
[gastonv@d5152011f ~]$ cat /proc/asound/cards
0 [AudioPCI ]: ENS1371 - Ensoniq AudioPCI
Ensoniq AudioPCI ENS1371 at 0xec00, irq 17
[Code]....
I have an Intel HDA audio card, and would like to record both my soundcard output and my microphone input at the same time. Getting to record my soundcard output (aka "Stereo Mix" or "What-U-Hear") is impossible for my soundcard without using the PulseAudio monitor, so I've set that up. Now I can record either the PulseAudio monitor for my soundcard, or the microphone input, but I don't know how to record both at the same time.
Currently, my .asoundrc looks like this:
Code:
pcm.pulse {
type pulse
}
ctl.pulse {
[code]....
So now I can choose either "pulse" as an input channel in, for example, Audacity or RecordMyDesktop applications (and then set in the PA volume control the actual channel which I want recorded), or choose "pulse_monitor" or "pulse_mic", in order to record either the PA monitor or the microphone, respectively.
I've read through the ALSA plugins reference, as well as the Asym and Dmix pages on ALSA wiki, but I'm still not sure how to put this together. I figured there should be a way either to route the microphone input into the ALSA output, and thus make the PA monitor "hear" what I speak into the microphone, or to make a completely new channel with both ALSA output and microphone input as "slaves", and then use that for recording.
How can I record text to speech output from festival to an audio file like mp3 or ogg?
My guess would be to pipe it to a command line recorder
i am not sure if this hardware section is the proper place to put this question but my problem is like this. I want to record headphone mic input in a local file every time my system boots up and broadcast the real-time audio or rather multicast it through local LAN
View 1 Replies View Relatedi am reading a database flat file with the "awk" command which has 4 fields separated by colon ":" .I want to show the output of these fields in a certain way that every record is numbered e.g
1.some text
2.some text
Is there a way to do this?
I recently got a real fancy unidirectional microphone, but when I go and use it, something strange happens. The mic is a mono mic, but when I record something, it appears to think it's stereo. Instead of having the same thing on both channels, it puts it all in the left channel, and silence on the right channel.
I tried fiddling with alsamixer. In capture (F4), adjusting the "Capture" device has an effect. However, it doesn't do anything useful. Lowering the right channel to zero does nothing, lowering the left channel to zero creates silence, and disabling the right channel creates silence.
In my System>Preferences>Sound (or pavucontrol, or what-have-you), there are "Analog Stereo Input" and "Analog Stereo Duplex" options there, but no "Analog Mono Input" or "Analog Stereo Output + Analog Mono Input" options (the latter being ideal).
Something interesting, though, if I do this:
Code:
arecord -r 96000 -D pulse -c 1 -vv -V mono /dev/null
it works correctly and outputs on both channels. But if I do
Code:
arecord -r 96000 -D pulse -c 2 -vv -V stereo /dev/null
then it's back to the same behavior. The VU meter on the second command shows the left channel going up and down as I make noise, but the right one always at zero.
The machine does have a crap internal mic built in to the screen bezel. If I do either of the above commands, they both work correctly. The stereo VU shows the same activity for both channels.
The machine is a Eee 1015PED, which has Intel integrated sound. Something probably insignificant is that the mic has a 1/4 inch mono plug, and to get it to plug into the Eee, I needed to get a 1/4 female to 1/8 male adapter. The mic's 1/4 plug is mono (duh) and therefore two-conductor, but the adapter I got is stereo, or three conductor (L, R, GND). I don't know if this is tripping something in hardware to tell it it's stereo or something.
I tried the mic on my desktop, which has a SB Audigy 4 (the non-pro flavor) and that has the "Analog Mono Input" function, so that works fine. Is there some way to turn this stereo input into mono input?