New answers tagged audio
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TiMidity++ supports both Windows (-iW) and ALSA (-iA) virtual MIDI interface creation.
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I'd suggest using the ALSA loopback driver. To load it:
sudo modprobe snd-aloop
This should add a new capture device to the output from arecord -L.
If you have only one physical soundcard in your computer, you can make the loopback soundcard the default for all apps by creating a file .asoundrc in your home folder with the following content:
...
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To decode from AAC to WAV, you can use FAAD2, which is a command-line utility.
faad -o output.wav input.aac
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I don't use Gnome3 so I can't check this, but according to this thread, you should be able to do the following:
Run dconf-editor
Navigate to org -> gnome -> desktop -> sound -> event-sound and deselect it. You may also want to deselect input feedback sounds if selected.
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See: Does DVI to HDMI carry audio?
So the answer seems to be that you're assuming the DVI port can throughput audio, which might not be true. If your graphics card has an HDMI port, use that. If not, get a card that does.
Another source: "The 9500GT does not support audio or have HDMI out. A DVI-D to HDMI connection carries no audio."
If you have a sound ...
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I agree with @Conn Darcy with one exception. Not all onboard sound cards are equal. Some cheap onboard sound cards rely on the CPU for processing. This may be the case here. The easy way to test would be disable (not mute) the onboard sound card and play your game. If the game runs fine, then its the onboard sound card. If this is the case, then you ...
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The GTX650Ti uses audio passthrough to provide sound. Basically, the sound card built in to your motherboard is used to process the sound and then it passes it onto the graphics card which outputs it via HDMI. So you are using the on board sound card. Most on board sound cards are fine and shouldn't cause you any lag issues so it is not that causing the ...
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It sounds like an issue I had in my past when the inccorect amount of stand offs were used. Be sure to review the installation of the required stand off points. Putting too many or too little will allow the board to run the risk of shorting out or being improperly grounded. This can cause immediate shutdow, to BSOD and even death of the motherboard.
Too ...
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Have a look at XmediaRecode for ripping the DVD to just audio, (select custom output and then audio only)
Not sure about creating a slide show, maybe this or use Windows Movie Maker
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When you talk about "efficiency", you probably mean the subjective audio quality one codec offers at a given bit rate, i.e. how humans perceive the quality of a sound under certain conditions.
Now, there are far too many codecs to be able to compare those. Today, the most relevant are:
MP3
AAC
Vorbis
In the future we might see Opus getting more exposure ...
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I've used MediaTomb as an DLNA Server under Xubuntu, because it is controlled with a webinterface it should work with KDE just as well.
MediaTomb installation description
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The first part of the question - related to re-enabling the disabled microphone was answered here.
So, I have a partial answer:
In the 'Sound/Recording' window, right-click and there's an option to see the disabled devices, which then can be enabled.
I had expected the internal microphone to be disabled automatically when an external one is connected. ...
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have you checked the components inside the machine Processor and Memory to make sure they are not loose?
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While you can connect the two PC's and play a movie on your laptop from your desktop, Video Games are a little difficult. mstsc automatically will play sounds from the remote machine on your local machine. So soundwise, you're set, Games however, are a very different story in my experience.
Directions
Run mstsc
Type in IP address of local laptop from ...
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I think this is due to shorting.
To find out,
1) Buy a mini jack cable (male on one end, female on other (see image below, they're very cheap)),
2) turn PC off,
3) plug the cable in to the faulting jack port,
4) turn PC on,
5) when OS loaded plug your audio into the new cable and see if the same issue persists.
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Use the volume mixer that is part of Windows 7. If you right-click on the speaker icon, select "Open Volume Mixer". A window will open showing you your sound levels for the the system device and all applications you have open. If for example you use VLC to listen to your audio files of movies, you can set the volume for that program to 80% of your master ...
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Here is an easy and cheap solution. Get a headphone splitter with built in volume knobs. Like this. Just lower the volume to about 80%. This means at 100% output from your PC, the stereo will only get 80%.
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First, have you checked your Skype audio settings? Skype has the final say as to where its audio/video feeds go. This is so that you can set your system audio to your speakers, such as a game or video, but have a Skype call sent to the headphones so that it remains private and doesn't affect the other applications.
Second, you can right click on the ...
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I think that you're telling the computer to only use the headphone for communications applications. When the headphones are plugged into the jack, the logic still outputs sound from applications like WMP and youTube to the speakers, because the headphone jack is reserved for communications applications like VoIP. I think that if you disable the headphones, ...
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I'm going to list the various troubleshooting steps I took and what the result was:
Restart Computer: No difference
Reinstalling my audio drivers (Realtek HD Audio): No difference
Tried with and without AMD HDMI Audio Drivers: No difference
Update Chipset and BIOS driver: No updates available
When the problem occurs try to disable/reenable AMD HDMI Audio ...
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You can change default device as well as default communication device to headphones. That should take care of it.
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This Gentoo wiki article might apply here : PulseAudio per-application volume control :
PulseAudio supports per-application volume control, but by default
this doesn't do much as you can only control these volumes from the
pulseaudio volume control utility. Meaning that in an application like
Audacious, when the output device is set to PulseAudio, ...
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There should be a file named setup.exe, click on that and follow the process through. After a reboot it usually works fine.
The reason is that what you downloaded is a zipped file, that has been extracted and what i assume you see is the folder with the installation data, it didn't actually install the driver yet. There are other ways to install a driver, ...
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According to the manual entry on aresample, you have to supply the resampler options in a different format:
The filter accepts the syntax [sample_rate:]resampler_options, where sample_rate expresses a sample rate and resampler_options is a list of key=value pairs, separated by ":".
That means you'd need to call it like this:
-af ...
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If you have Mac OS X, you can use LineIn to route the microphone input to your headphone jack. You could then have a cable going from the headphone jack on your laptop to the input jack on your desktop.
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If you want to play your synth through your Mac audio, Quick Time would not work because of huge delay (latency). LineIn's latency is smaller but still too strong.
So I need to run Logic which is not very practical but it is the only solution I have found so far. This is VERY odd because I am quite certain that it would be very easy to add "Audio Pass ...
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This is a complementary answer made to record what I consider to be the meaning of the other answers so far. There are different ideas floating here, maybe because my question was too general or vague. I have edited it to clarify, but the bad is done.
When a video file is the input, check (like this or this) what kind of audio it contains (for the purpose ...
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Regarding the most efficient audio format
Overall, I would probably pick AAC, because it is widely supported, supports a wide range of bitrates and usually beats competitors at any bitrate. Furthermore, AAC has a low bitrate mode called HE-AAC which employs some sophisticated algorithms to reproduce high frequencies and stereo in a very bandwidth-preserving ...
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First. If you really want to know how MP3 works, check this article about the theory of MP3 by Rassol Raissi
The article dates from 2002, may look outdated, but the author clearly explains the difference between sampling frequency and bit rate. Those are two totally different concepts.
Second. MP3 is a protocol. It's not an algorithm. Every implementation ...
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For those interested here is the final working code :
#!/bin/bash
#
# Batch conversion of audio extraction from video
echo $BASH_VERSION
FOLDER_SRC="/home/J/Music/TestBatchConv"
shopt -s globstar
for myvid in "$FOLDER_SRC"/**
do
echo "$myvid"
avconv -i "$myvid" -acodec libmp3lame "${myvid%.*}.mp3" 2>/dev/null || echo "$myvid is not a valid file."
...
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When a certain file (mp4, flv, etc) has a 95 kbps audio bitrate - does
it make sense outputing to a higher bitrate when converting to mp3 or
other format (be it lossy or not)?
It may make sense, since we're talking bits per second in different formats and not sampling frequency.
As an extreme case, suppose you have an uncompressed raw file with 16 ...
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It doesn't make sense to re-encode audio to a higher bitrate, but the bitrate might have to be somewhat higher if you want to reduce a further degradation in quality.
You should avoid transcoding audio whenever possible.
If you need to change the video format, you may be able to keep the audio in the same encoding.
For instance if you use the ffmpeg ...
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Seems you're using an old version of file where MKV files weren't detected properly. You can use this ~/.magic file to have file identify MKV.
But actually, the most reliable way to check for valid video files would be to just run avconv and see if it fails or not. Here's what you can do:
#!/bin/bash
FOLDER_SRC="/home/J/Music/TestBatchConv"
find ...
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First it's correct that you don't get more information from up sampling. But combining up sampling with a low pass (or interpolation) filter will get you a smoother curve. Passing this to the stereo should result in less noise produced from the stereo trying to reproduce the noise given by the original low sampling rate.
The important factor here is that ...
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There are some excellent technical descriptions of why this is a bad idea in this thread; to offer a different perspective, imagine that every time you make a lossy-compressed audio file (MP3, OGG, AAC), it's like dubbing a cassette tape. Even if you buy the most robust, highest-quality tape you can buy, every time you dub it, all that does is minimize the ...
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Yes, it might actually make sense if you are being forced to change formats.
If you have a file with 95kbps in a highly efficient format, to retain the same quality, a relatively inefficient format as mp3 needs a higher bitrate.
Of course you will never get anything back that was lost in the first place. On the contrary, encoding as mp3 will reduce the ...
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You can't "improve" the signal by re-encoding the output into another lossy format (mp3 etc.). It will always be worse than the original.
If you must re-encode it, the best result you can achieve is the same quality by choosing a losless codec like FLAC or ALAC. Or even uncompressed formats like WAV.
If there's no other source for your file you should keep ...
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By increasing the bitrate you won't have an higher sound quality.
Think about it this way: when it was converted from the original media (let's say a CD) it was compressed to fit the "content" in a smaller "box", and by doing so an amount of data has been lost (you may want to read about lossy and lossless formats).
If you subsequently increase the bitrate, ...
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In the general case this will not usually result in higher quality audio. The basic reason being that you cannot manufacture sounds that aren't there in the original file.
In the best case the only result will be, as you suggest, larger files.
In the worst case the files could even be of worse quality as the second lossy encoder is tying to encode the ...
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You could use Audacity. It will give you a visual representation of the volume. It is open source and very good.
http://audacity.sourceforge.net/
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This seems to be solved at this arch thread. You basically streams your alsa output serving and transcoding it to mp3 at the same time with ffmpg. To achieve so you need to:
Load a kernel module, virtual soundcard.
Tell Alsa to use this soundcard with a config file.
Tell ffmpg to listen to this soundcard with another config file.
Start the server.
Then ...
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Running the file through AsfBin might help. Add the file to the list of input files, make sure that a valid destination is set, and click the Cut/Copy/Join button.
If that doesn't help then I suggest that you extract the individual audio and video stream from the file ('demux') and that you check if the audio stream isn't just all silence.
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Loosely speaking, the audio signal path in the sound card looks like this:
Microphone -> Preamp -> | Vol.Cntrl -> PowerAmp -> Speaker
D/A Conv-> |
LINE-IN -> |
|
V
LineOut
So basically, The LineOut is a signal output (high impedance, low ...
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http://www.streamwhatyouhear.com
It's even free, and it should do exactly what you want.
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The only component that you really have much to worry about is a hard drive. If you have an SSD the point is moot.
Otherwise, a foot would be more than enough space (This was about average for poorly shielded speakers and PCs / CRT monitors). Although, with decent speakers I have seen more than a few speakers sitting on the PC and there has been no ...
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You should have no issues with them right next to each other. All your devices are well shielded. In addition to that, home based electrical devices do not typically generate a large or powerful magnetic field. As for proof, well... There are plenty of HTPC, DVRs, etc that are more than 1000 watts that have hard drives inside them.
As for sound, I ...
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The best solution I've had is by using a media player that supports selecting the output audio device.
Personally I use VLC, and the option still seems hidden.
In the preferences dialog, with all options enabled, it's under Audio > Output modules > DirectX.
I just keep it set to my HDMI connection and my speakers as the default audio device.
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Windows will output audio to the default device automatically. Some applications have software options to output sound to a device other than the default device. A good example of this would be running a game on your computer and using Ventrilo to output other players audio to the headset.
You may be able to play with the "default communication device" ...
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I just accidentally found a lame and hacky but working way to solve the issue - plug the headphones halfway, instead of fully, to the port :)
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A 2 dollar jack splitter solved this problem for me.
http://www.xxion.co.uk/images/uploads/Jack_Splitter_Gold.jpg
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