Devices and playback

After setting my interface (M-audio Firewire solo) as the device in the audio preferences I found that I can only hear anything through headphones connected to the interface or by setting the device back to 'built in microphone'. Is there a way of setting it to stop sending the sound through the interface?
Cheers

>importing them into Adobe
Adobe is a company... which program are you using?
Also, iphone video has problems due to a variable frame rate
- A possible fix in message #22 http://forums.adobe.com/thread/934466
For CS5 and later, the easy way to insure that your video and your project match
See 2nd post for picture of NEW ITEM process http://forums.adobe.com/message/3776153
-and a FAQ on sequence setting http://forums.adobe.com/message/3804341
The tutorial list in message #3 http://forums.adobe.com/message/2276578 may help

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  • A quick primer on audio drivers, devices, and latency

    This information has come from Durin, Adobe staffer:
    Hi everyone,
    A  common question that comes up in these forums over and over has to do  with recording latency, audio drivers, and device formats.  I'm going to  provide a brief overview of the different types of devices, how they  interface with the computer and Audition, and steps to maximize  performance and minimize the latency inherent in computer audio.
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    Sample: The value of each individual bit of audio digitized by the audio  device.  Typically, the audio device measures the incoming signal 44,100  or 48,000 times every second.
    Buffer Size: The  "bucket" where samples are placed before being passed to the  destination.  An audio application will collect a buffers-worth of  samples before feeding it to the audio device for playback.  An audio  device will collect a buffers-worth of samples before feeding it to the  audio device when recording.  Buffers are typically measured in Samples  (command values being 64, 128, 512, 1024, 2048...) or milliseconds which  is simply a calculation based on the device sample rate and buffer  size.
    Latency: The time span that occurs between  providing an input signal into an audio device (through a microphone,  keyboard, guitar input, etc) and when each buffers-worth of that signal  is provided to the audio application.  It also refers to the other  direction, where the output audio signal is sent from the audio  application to the audio device for playback.  When recording while  monitoring, the overall perceived latency can often be double the device  buffer size.
    ASIO, MME, CoreAudio: These are audio driver models, which simply specify the manner in which an audio application and audio device communicate.  Apple Mac systems use CoreAudio almost exclusively which provides for low buffer sizes and the ability  to mix and match different devices (called an Aggregate Device.)  MME  and ASIO are mostly Windows-exclusive driver models, and provide  different methods of communicating between application and device.  MME drivers allow the operating system itself to act as a go-between and  are generally slower as they rely upon higher buffer sizes and have to  pass through multiple processes on the computer before being sent to the  audio device.  ASIO drivers provide an audio  application direct communication with the hardware, bypassing the  operating system.  This allows for much lower latency while being  limited in an applications ability to access multiple devices  simultaneously, or share a device channel with another application.
    Dropouts: Missing  audio data as a result of being unable to process an audio stream fast  enough to keep up with the buffer size.  Generally, dropouts occur when  an audio application cannot process effects and mix tracks together  quickly enough to fill the device buffer, or when the audio device is  trying to send audio data to the application more quickly than it can  handle it.  (Remember when Lucy and Ethel were working at the chocolate  factory and the machine sped up to the point where they were dropping  chocolates all over the place?  Pretend the chocolates were samples,  Lucy and Ethel were the audio application, and the chocolate machine is  the audio device/driver, and you'll have a pretty good visualization of  how this works.)
    Typically, latency is not a problem if  you're simply playing back existing audio (you might experience a very  slight delay between pressing PLAY and when audio is heard through your  speakers) or recording to disk without monitoring existing audio tracks  since precise timing is not crucial in these conditions.  However, when  trying to play along with a drum track, or sing a harmony to an existing  track, or overdub narration to a video, latency becomes a factor since  our ears are far more sensitive to timing issues than our other senses.   If a bass guitar track is not precisely aligned with the drums, it  quickly sounds sloppy.  Therefore, we need to attempt to reduce latency  as much as possible for these situations.  If we simply set our Buffer  Size parameter as low as it will go, we're likely to experience dropouts  - especially if we have some tracks configured with audio effects which  require additional processing and contribute their own latency to the  chain.  Dropouts are annoying but not destructive during playback, but  if dropouts occur on the recording stream, it means you're losing data  and your recording will never sound right - the data is simply lost.   Obviously, this is not good.
    Latency under 40ms is  generally considered within the range of reasonable for recording.  Some  folks can hear even this and it affects their ability to play, but most  people find this unnoticeable or tolerable.  We can calculate our  approximate desired buffer size with this formula:
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    Effects: A  common cause of playback latency is the use of effects.  As your audio  stream passes through an effect, it takes time for the computer to  perform the calculations to modify that signal.  Each effect in a chain  introduces its own amount of latency before the chunk of audio even  reaches the point where the audio application passes it to the audio  device and starts to fill up the buffer.  Audition and other DAWs  attempt to address this through "latency compensation" routines which  introduce a bit more latency when you first press play as they process  several seconds of audio ahead of time before beginning to stream those  chunks to the audio driver.  In some cases, however, the effects may be  so intensive that the CPU simply isn't processing the math fast enough.   With Audition, you can "freeze" or pre-render these tracks by clicking  the small lightning bolt button visible in the Effects Rack with that  track selected.  This performs a background render of that track, which  automatically updates if you make any changes to the track or effect  parameters, so that instead of calculating all those changes on-the-fly,  it simply needs to stream back a plain old audio file which requires  much fewer system resources.  You may also choose to disable certain  effects, or temporarily replace them with alternatives which may not  sound exactly like what you want for your final mix, but which  adequately simulate the desired effect for the purpose of recording.   (You might replace the CPU-intensive Full Reverb effect with the  lightweight Studio Reverb effect, for example.  Full Reverb effect is  mathematically far more accurate and realistic, but Studio Reverb can  provide that quick "body" you might want when monitoring vocals, for  example.)  You can also just disable the effects for a track or clip  while recording, and turn them on later.
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    There is one point in the above that needed a little clarification, relating to USB mics:
    _durin_ wrote:
     If  USB microphones are your only option, then I would recommend making  certain you purchase a high-quality one and have an equally high-quality  playback device.
    If you are going to spend that much, then you'd be better off putting a little more money into an  external device with a proper mic pre, and a little less money by not  bothering with a USB mic at all, and just getting a 'normal' condensor  mic. It's true to say that over the years, the USB mic class of  recording device has caused more trouble than any other, regardless.
    You  should also be aware that if you find a USB mic offering ASIO support,  then unless it's got a headphone socket on it as well then you aren't  going to be able to monitor what you record if you use it in its native  ASIO mode. This is because your computer can only cope with one ASIO device in the system - that's all the spec allows. What you can do with most ASIO hardware though is share multiple streams (if the  device has multiple inputs and outputs) between different software.
    Seriously, USB mics are more trouble than they're worth.

  • Is it possible to do the audio capture and playback simultaneously?

    In short :
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    In detail:
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  • SbLive! Value Devices - Midi playback differen

    Creative SbLi've! Value.
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    2. SbLi've! SynthB(DCEO) General MIDI
    3. SbLi've! SWSynth (DCEO) General MIDI
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    charlesh wrote:
    Creative SbLi've! Value.
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    2. SbLi've! SynthB(DCEO) General MIDI
    3. SbLi've! SWSynth (DCEO) General MIDI
    4. SbLi've! MIDI Port (DCEO) General MIDI
    5. Microsoft GS Wavetable SW Synth
    6. Kontakt Player Silver (Sibelius 3 music notation program)
    All test OK for playback. But SynthA (item ) is much louder than the others. Also playing MIDI scores SynthA does NOT read some text marks, like the volume (ff, pp) instead plays volume uniformly loud ignoring volume (velocity) notations but accepts for ex. tempo notation.
    What is the difference between the devices (for example SynthA vs. SynthB)? Where can I find a good explanation how these devices suppose to work, how SbLi've reads Midi instructions in scores etc.
    In playing Midi scores I wish to use my own soundfonts (sf2) which are generated by downloading existing ones and modifying them. Is there a limit for their size in SbLi've! Value board?
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    I think the SynthA and SynthB should behave similarly because you can select one or the other (in SB AudioHQ - SoundFont - Options - Soundfont Device). I haven't been playing MIDI files for sometime, and it's good you mentioned that it doesn't read some text marks (I didn't know about this... but I don't compose MIDI music... so it doesn't matter that much to me)
    The SWSynth is Creative's software synthesizer, I suppose (the SynthA and SynthB are hardware synthesizers -- check them using DXDiag.exe which will list these devices and their capabilities).
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    Now you can use Logic's Preferences > Devices > Core Audio >Device list to see your Aggregate Device. I’ve named mine “Snowy Line Out” to trigger my recognition of what it contains. Another one is called “Snowball + Line + Digital” and it has both line and digital outputs aggregated to the Snowball mic input.
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    I hope this gets you through the frustration of not knowing what to do when confronted with the lack of clear docs and complete steps to get a USB mic or other seemingly opaque input issue. Feel free to add to this so others in my former state of bewilderment can gain understandings.
    If Steve Jobs were to encounter the level of opacity, ambiguousness and frustration that setting up a simple USB microphone can create in Logic, you can bet that the several programs and their intimate interdependency would be either automated or fully explained in the documentations that followed his fiery analysis of the issue.

    I don't know of another way to get the Blue Snowball USB mic to show up in Logic. If you can do it without having to go through the Aggregate Device route, feel free to post here how to get that to happen.
    I didn't have a different mic available when I started trying to do what seemed to be the simplest of things -- record and play back an audio track -- and nothing else seemed to work.
    There are a lot of Aggregate Device threads around here, many associated with Blue Snowball, Blue Snowflake and other things, but none of them laid out the principles of how Aggregate Device Setup needs to work or what to do to force it into compliance.
    As I said, the clues in the Logic docs and in-mac docs are minimalist, at best, none of them bringing instant head slaps of success readily.

  • Sharing one iTunes account between 3 devices and a Time Machine

    I have a iphone 4S
    i have an Imac
    I have a Time machine
    I have a macbook
    I have Apple TV - hard wired to the TV
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