How to choose between different audio output in different programs

I want to choose between the audio outputs in different programs.
For example; I want to hear spotify/iTunes with my plugged-in speakers and skype with the build-in/intern speakers.
It is possiple to control this on Windows but can I do it on Mac?
Does anyone know an app for this problem?
Thanks for help!

I'd assume you'd want to set the sequence to "slur" notes and then just calculate the notes between a top & bottom value based on a float and generate NOTE_ON messages appropriately.
The pitch bend is designed to emulate / support the bend wheel found on sound keyboards...and it just modifies the pitch associated with a given note. A pitch bend of +2 would turn an A into B, etc...
So, you'd want to do...
SET_SLUR (however you do that, I'm not a MIDI expert)
pitch = bottom_pitch + ((top_pitch-bottom_pitch) * sliderValue);
NOTE_ON(pitch)

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Different Audio Outputs Logic Studio 8

Hello. I was wondering if apple was able to do this, because that would save a lot of plugging and unplugging of cables and such. Is it possible to select different audio outputs for different programs that you are using on the imac? I run logic studio 8 with a m-audio firewire 410 external soundcard with a couple of monitor speakers. I also have a set of z-5500 connected to my imac via the digital output cable. Is there a way where when i use logic studio 8, the firewire soundcard is used. and for all other purposes, such as watching a movie or listening to music, the digital output is used? Is there a program that can control this? if this is possible. it would help me out A LOT! THANKS

Hello ecstasurf
That's a great question?
Go to: "Applications / Utilities" and open "Audio MIDI Setup" from there you maybe able to switch between the FireWire and the Digital/Analog outputs. I can't give you much more help with that, I don't have an audio FireWire setup so you'll have to play around with the Audio Output Source there.
Note: If you like, you can drag a copy of the Audio MIDI Setup to your dock for quick and easy access.
Dennis

HT1222 my iphone 4S no longer plays music output to speaker after updating to version 6.1.3. I am unable to reset the phone to an earlier version! How do I get my audio output to speaker back? (music, audible, youtube, netflix))

my iphone 4S no longer plays music output to speaker after updating to version 6.1.3. I am unable to reset the phone to an earlier version! How do I get my audio output to speaker back? (music, audible, youtube, netflix)

So you have business contacts that were lost... do you not have these on a Database in some shape or form? Sounds like you need to invest in a RAID1 backup setup for your computer so you dont have this issue in the future.
As for repairing your problem, sounds like you are going to need to start from scratch at this point or if the data is still on your phone look into a program that can take the information from the phone onto iTunes.

How can i install my audio output device back

my audio is not working it says "No Audio Ouput Device installed" but it was working perfectly fine the day before. PLease Help

Re: how can i install my audio output device back
earlier its working good but not now

How to connect Digital Optical Audio output iMac with Receiver Input

I have connected iMac with onkyo hts-3300 home theater system receiver through optical audio TosLink Cable.( http://www.amazon.com/Digital-Optical-Audio-TosLink-Cable/dp/B001ZU3G2I/ref=sr12?ie=UTF8&qid=1292865309&sr=8-2 ). Unfortunately there is no sound coming out from the Home theater system. I have verified the Receiver optical input is working fine by connecting with my TV optical audio digital out put and the it's working great. The only difference between TV and iMac is, on my TV, it has optical audio digital output and I can use this cable (http://www.amazon.com/Tripp-Lite-A102-02M-Digital-connectors/dp/B0002WPREU/ref=s r11?s=electronics&ie=UTF8&qid=1297320792&sr=1-1) to make it work. I can also verified my optical audio TosLink Cable working because I can see the red light coming out from the cable. So the only problem could be from iMac. I doubt I need to change the additional setting to configure iMac to use external optical audio setting instead of the buit in audio setting. Hoe can I able to do it? or my iMac(IMAC 21.5"/3.06/2X2GB/500GB/9400M-ITP
MB950ZP/A) doesn't support optical digital audio out put? I have follow this thread http://discussions.apple.com/thread.jspa?messageID=12769423&#12769423 and to my surprise which doesn't work out for me. I doubt that i need to change some setting, but when I got to setting>> sound I cannot change any setting. I was quite depressed aldy. Can someone answer my question as i m a newbie here. Thanks a lot

Thank you very much for your reply.
@Király : May i know how to Open Audio Midi Setup? Is this Audio Midi Setup is in the locatated in the Setting in Mac Snow Leapord OS? or how to locate this "Audio Midi Setup " in Mac enviroment. Is it possible to open this Audio Midi Setup in windows 7 enviroment also? I Just want to get the audio from my iMac in Hts-3300 onkyo home theatre. YOur help is much appreciated
@rkaufmann87 : Thank you very much for your sharing about this http://www.apple.com/airportexpress/ connection. for me what i understand is the digital audio output in airportexpress is the same as the audio out put in iMac in terms of details sepcs. So this airportexpress will only help you connect wirelessly to home theatre system, rather than waired connection. Other than that nothig much difference. So there is no point for me to spend extra bucks to attain the wireless connection. Unless the airportexpress have optical digital input, which i believe not. Correct me if I m worng, thanks

Is there no way to direct an AIR app's audio to different audio output devices?

I'm needing to make a couple of our AIR apps select certain audio devices to send their audio output to, and last year, somebody mentioned possibly using native extensions to do this. I had to start working on other things for a while, but I've come back to this (for the moment), and I've basically learned how to create native extensions.
But so far, they don't seem to provide me any special access to anything going on under the hood of an AIR app. When you stream live audio in through RTMP or RTMFP, the details of the audio are hidden from you. There doesn't appear to be any place where you can actually access the audio bytes that are being sent over the wire. You can specify a certain SoundTransform object for the NetStream to use, but that class is final and a vanilla Flash construct, so even if it may have a way to access the audio bytes, they're not acessible at all.
It really looks like, when a NetStream object starts streaming in live audio, there is no way under the sun to access the audio bytes and direct them to a certain audio output device, with or without extensions. Is this so? It would be unfortunate if Adobe didn't allow this, even through native extensions. Thanks.

Couple quick thoughts..
Limits are important. There's a large amount of work done to keep a clear separation between a users decision and a developers decision. For instance, you can't change a users default audio output because that clearly violates a users decision. That said, here's a whole mess of people debating the issue and actually eventually an undocumented COM API you can find that will actually let you do it in Vista/Win7 (but not 8), however don't expect that to be a 'good thing to do':
http://social.microsoft.com/Forums/en-US/9ebd7ad6-a460-4a28-9de9-2af63fd4a13e/programatica lly-setting-the-default-playback-device-and-recording-device?forum=windowspro-audiodevelop ment
You might want to get a beverage before reading that.
Aside that, RTMP is secured, which is why you can't gain access to the bytes. The entire purpose of it is to assure media developers that their media can't easily be copied/stolen. Adobe doesn't document their method of securing streams but they do publish the RTMP specification. It's entirely possible for you to create your own RTMP server and encryption method while delivering a RTMP stream to custom client that you code yourself which can read the stream. Here's the spec:
http://www.adobe.com/devnet/rtmp.html
So I'd say it would be unfortunate if Adobe did give you access to the bytes. You might as well just hand users a mp3 of whatever song is playing because they would steal media left and right.
That isn't to say you can't use a multitude of ways of getting the data going across the default output device. Plenty of apps (fraps/etc) capture video and mixed audio streams. But Adobe isn't going to give you the keys to the kingdom and let you decrypt their protected media protocol.

2 Tracks, 2 different audio outputs:

I have two tracks, exactly the same. However, one track (track 1) has a metronome click, the other does not (track 2). My laptop is connected to a Mackie Pro FX12 mixer. I am connecting my laptop to the mixer via the laptop audio output jack connecting to a tape input on the mixer. I would like track 2 (no metronome) to run through audio speakers and track 1 (metronome) to run through headphones. Both the speaker and headphone outputs are on the mixer. What do I need to do with Logic Express to make certain that the tracks are run through the correct output?

I'm having some problems with this as well. I want to have one track playing on one output (headphones) and then another track playing on another output (speakers). I have panned each track, so one is on full left and the other on full right. But I don't understand what to do now? How do I route each track to a different output? I don't know how to do this.
Many thanks in advance for your help.

Need your help choosing between 2 audio/midi interface for logic

Hi, new member here. I just signed-up just to ask for your recommendations.
I am looking to buy either the Cakewalk UA-4fx (http://www.sweetwater.com/store/detail/UA4FX) or Tascam US-122mkII (http://www.sweetwater.com/store/detail/US122mk2).
Do you have any advice on which one to get (or a third alternative if you know something that's similar but better)?
I'm mainly looking to record from my Roland Fantom X keyboard to my Macbook (using Logic Pro). The effects/filters on the Cakewalk device is not that important to me (since I can use software effects), but the quality of sound is the most important. Also, ease-of-use and compatibility with Logic Pro is important. The device also must include MIDI-in/MIDI-out in addition to 1/4 inch direct line in for my synth. (I plan on using Midi-In/Midi-out with MainStage in Logic Studio). Not looking to spend too much more than $100.
I think I read in a review that even though the Cakewalk UA-4fx has 2 inputs, it can only record 1 at a time. Is the same true of the Tascam or can it record 2 simultaneous tracks?
But in short, sound quality AND compatibility with Logic Pro and MainStage are the main concerns. Does anyone have advice on choosing between the 2 or a third option if you know of one?
Thanks in advance!

Hello -
It's been suggested to avoid, if possible, USB-based audio devices. My humble suggestion? I'd consider MOTU products. I own three MOTU audio devices (2408-MK3, 828-MK3 & 828-MK2) and all work just fine. The 828-MK3 has both midi and audio inputs & outputs (including ADAT inputs/outputs).
Other people will, no doubt, suggest audio devices from other companies. All probably get the job done well. But if you're thinking about recording a lot of instruments at the same time, it is my understanding that it is best to avoid USB-based devices.
Good luck with your search and your decision.

2 Different audio outputs

Is there a way to configure my mac to play iTunes out of the headphone jack and play all other programs (like safari) through my bluetooth headset? I'm a beginning DJ and was wondering if this could be done. What steps will I need to take to do what I want? New sound card?

I'm having some problems with this as well. I want to have one track playing on one output (headphones) and then another track playing on another output (speakers). I have panned each track, so one is on full left and the other on full right. But I don't understand what to do now? How do I route each track to a different output? I don't know how to do this.
Many thanks in advance for your help.

How do I switch Playback Audio Output ?

I see activity on the audio meters but FNCE is not communicating with audio device that I have selected on System Preferences Sound.
How can I manually tell FNCE which device to use for AUDIO OUTPUT ?
Please help

What device are you trying to use? Specifics always help.

How to stream Logic's audio output live via internet

Hi there, I'm looking for a way to stream the audio output from Logic (or any other DAW) from one recording studio via internet to another studio… not for remote recording, just for the producer for "creative monitoring", because he's out of town… so the audio quality doesn't need to be top notch. (Talkback would e.g. be via Skype.)
First question: is there a way to do it directly within the Logic DAW, i.e. from one Logic app to the other (remote) Logic app? I.e. just a "mirror" of the audio output (as stereo) that Logic will send directly to a remote host? I.e. a built-in Logic functionality? (I couldn't find anything in the manual.)
Second question: system-wise I assume it would be possible to do it by simply routing the output channels through Soundflower or JackOSX, and then into an application like Skype, where you would have to select the respective Soundflower/JackOSX channels as input.
One problem: the whole thing shouldn't interfere with the regular recording and monitoring setup in the originating studio. So maybe it would be best to simply have an additional monitor output from the hardware console or soundcard going by audio cable into a second computer, which will use the incoming audio as Skype input.
In a similar vein: is there a way to create a second pair of stereo outs in Logic, which functions as a direct audio "mirror"? It means that I would have the main stereo out signal on channels 1 & 2 (as always), but then also have the exact same signal on output channels 3 & 4 at the same time, which I could route to the soundcard's 3&4 outputs or through Soundflower into Skype?
Thanks in advance for your help.

Thank you, I think that answered my question. So it would go like this:
(1) All recording tracks plus playback tracks are routed not to stereo master output, but to a pre-master bus, e.g. bus 1.
(2) The pre-master bus (bus 1) output is then routed to stereo main out (1 & 2).
(3) On bus 2 (aux master) I select input from bus 1 (the pre-master bus), but route aux master bus output to stereo outputs 3 & 4.
(4) With outputs 3 & 4 I could then go directly into Skype, either on the same machine, or from the sound card 3&4 outputs into the input of a second computer running Skype.
I'll add an image in the follow-up post.

How to play ultrasonic to audio output

I'm monitoring an ultrasonic transducer operating in the 150Khz range and want to phase shift the sound to audio that can be played over the PC speaker output channel. The signal is very much like white noise with ocasional spikes that are very short in duration, around 100 microseconds.
I'm sampling the input at 500k samples / second so I need a way to not only shift the noise to the audio range but also not miss the spikes when resampled down to the ~40k samples / second for the audio output.
Any suggestions would be greatly appreciated.
thanks,
t

First, Thanks to all for the quick replies.
I'm monitoring the signal in near real-time, so playback at a slower speed isn't an option.
What frequency should the audio (output) sine wave be? What should the duration of the output spike be? Does the timing of the start of the output spike need to preserve the phase relationship to the input sine?
In the past, this has been done in HW and just output to a small 3 inch speaker so the preservation of the original signal isn't critical but being able to hear the pops is the desired result. I'm trying to get more information about the analog circuit that accomplished this.
The transducer itself is tuned for a frequency response around 155KHz. The source is very low power and already amplified 60db before getting captured on the NI device. The ambient nominal noise level is tuned in HW to be around 5 milivolts.
I've got a lot of latitude on the ouput. The duration of the spike needs to be sufficiently long just be audible and there is no sine to be concerned about phase. I'm processnig the signal at 10Hz.
I have taken a couple swipes at this problem. The first hurdle is that the signal is so low that once you amplify it to be audible, it's difficult to hear the spikes from the white noise. A spke may be as low 15 milivolts. The second is that when resampling for the audio, the spike can be skipped over completely.
My current approach is to first multiply the ultrasonic signal with a low frequency sine wave and then use a bandpass filter to try to filter out a harmonic. This seems to help improve the signal to noise ratio. The second step is to inspect every sample and dynamically amplify and widen each based on the level in order to make the spikes more pronounce. Finally, I resample the signal to meet the audio parameters and continuously play the result through the system speakers.
I'm not an EE so please excuse if my attempt to explain is less than optimal
Thanks,
t

How do I make two audio clips from different mics sound the same in premiere?

I have a video I am working on. we used two different mic and camera options. One is a lavaliere mic that was too hot and picked up a lot of fuzz, and the other was an on board mic that was just a little echoey from the office walls. I found that some clips were better on different versions. so I want to put the two different sets next to each other in video and make the sound be the exact same. That way no one notices the differences in audio.
Any help is appreciated.

Hi. Although not perfect, I'd add an audio mixer effect and adjust the highs, mids & lows on both of them to help their audio, respectively, sound more similar. Often when I shoot, my Boom mike (backup) is much fuller sounding than the lavs. are. Lavs are inherently lacking in the low end, and the boom is often too hot on the low end. So I would add Audio Mixer effect, and lower the low end on the boom slightly, and highten the low and mids just a bit on the lavs. Even taking a little highs off the lavs makes it sound a bit more like the boom.
As far as echo is concerned, you probably can't do to much about that unless you have an external graphic equalizer/limiter. Lot's of times I just add some music in the background, which helps a lot to hide echo from the mics.
Finally if you tape/record/film in a location that has lots of background noise like, background people, cars or rain or wind or general white noise "hiss", then you should record a minute or two of the location without any dialogue. This is is known as "Wala" recording. You can add a little of this to your whole track, or just on the mike track that doesn't seem to pick up the ambient sound, thus making it sound more like your other mike.
I hope this helps.

HT3625 My Macbook Air does not have this "Use audio for:" option, as it has been excluded from Airs for whatever reason. How do I change my audio output to input without this?

If I have to buy an audio-usb converter let me know.
Although if this is an option exclusive to Pros and I do, then this is a little ridiculous Mr. Ghost of Steve Jobs.
Especially considering how Airs don't have their own section here on the forums
Thank you for whoever responds! All the idiot "how to" links on Google have me a little frustrated

Nevermind, posted this in Air section
It didn't give me that option initially.

How to choose between CBR and 2-Pass VBR ?

Hi,
I know alot of virtual ink has flowed about this topic, but I couldnt find the answer I need.
I'm compressing an 83minute feature film to DVD. Source material is 10bit Uncompressed YUV from PAL DigiBeta, sent directly from FCP to Compressor. Compressor is at version 2.0.1, and all ProApp and QT updates are applied.
Compressing at a CBR (ie: 1pass) with a data rate of 7.4Mbps gives a .m2v file size of about 3.3GB and takes about 3hours.
Compressing at 2-Pass VBR with a max data rate of 7.4Mbps gives a file size of about 3.9GB and takes 12 hours+.
Shouldnt the two pass be smaller given the same max data rate?
When I compare the two visually, I cannot tell any difference between them. I'm looking at areas with alot of motion, or with fine detail, or camera movement, but they look about the same.
How to decide which to use?
Many thanks....
G5 Dual2.0 (PCI-X), 4.5GB RAM, 23" Cinema Display, PB G4 Mac OS X (10.4.6)

Hi There
CBR compression is generally better quality - especially out of Compressor 2 - but usually a higher file size that VBR - can't explain your result there. Check fades, dissolves, small text and camera movement for comparsion between the two mpegs. If you still can't tell the difference then use the bigger one if you have the room, if you don't use the smaller one. If the larger file is better but you're worried about disc space then build your disc using the lager one anyway. If after your created the VIDEO_TS file it still too big use DVD2ONE or DVD Remaster to shrink it back down - I doubt you'd see any visiual difference.
Hope that helps
Cheers
B

How to choose between different audio output in different programs

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