T61 audio recording device

Hi,
I have a T61, I'm trying to set the audio recording device to be what's playing out of the speakers. I only see microphone. I've checked the "Show Disabled/Disconnected devices", and there's nothing else under there. Running windows 7.
Thanks
Moderator Edit: Topic title adjusted from T60 to T61 for clarity
Solved!
Go to Solution.

http://software.muzychenko.net/eng/vac.html
this is software that many people use to overcome this issue, and it is not free.
Regards,
Jin Li
May this year, be the year of 'DO'!
I am a volunteer, and not a paid staff of Lenovo or Microsoft

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An audio recording device cannot be found

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Can you post back with the full Model No. and Product No. of your g6 series notebook ( from the service tag underneath your notebook ) - see Here for a further explanation.
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My external audio recording device only records to the right channel. How do I send the sound in the right channel thru the left one in Final Cut to get a full stereo sound?

My external audio recording device only records to the right channel. How do I duplicate the sound in the right channel and run it to the left one in Final Cut to get full stereo sound?
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Hi, thanks in advance for any help!
I'm having difficulty recording on Windows Vista. Having installed my soundcard recently from a previous computer, I downloaded the relavent Vista Driver: "Creative Sound Blaster Audigy series Vista Driver 2.2.0002" (6 Mar 07). I have not installed the beta version released 02 Jun 08 (perhaps someone will recommend i try that).
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Copied from the Video forum - (related to video operation under Vista)
I have used the follwoing procedure to install a working configuration of the Audigy 2 ZS Video Editor in a clean install of Vista Ultimate.
() Download from Creative Labs (US) site: http://us.creative.com/support/downloads/ the following files:
(a) Creative Sound Blaster USB Series Windows Vista Driver .00.0060 - USBVS_PCDRV_LB__00_0060.exe (from the Vista section)
(b) VidCap for Creative Sound Blaster Audigy 2 ZS Video Editor .00.000 - A2VX_PCVDE_L9__00_000.exe (from the WinXP section)
(2) Use WinZip or WinRAR to open the file (b) and extract the folders \DrvInst and \Vidcap into your Vista Downloads folder.
(3) Connect the Video Editor to the power supply and a video source; connect the USB cable to your PC. Ignore the "Found NEw Hardware" dialog that pops up.
(4) Doubleclick on file (a) to install the Creative Sound Blaster USB Series Windows Vista Driver, and follow prompts as required; accept all defaults. After install is complete, let the PC reboot.
(5) After rebooting, in the "Found New Hardware" dialog that appears, select the options to search for a driver, don't go online to search, browse for a driver and point the search at the \DrvInst folder you extracted in (2) above. The USB video driver will be installed from the \DrvInst\Video\Driver\Win2k.
(6) After the video driver is installed, doubleclick on the Setup.exe file in the \Vidcap folder you extracted in (2) above to install the Vidcap application.
(7) Setup the options in Vidcap, and run a test recording.
Happy capturing under Vista.Message Edited by Dale-CL on 04-8-2007 :3 AM

How can I record an audio file directly from a digital recording device directly to the iMac?

How can I record audio directly to my iMac from a digital recoding device?

What digital recording device?

Recording audio from device on server

I have some really nice audio recording gear hooked up to my computer, and I got a neat idea. I want to turn my PC into a web server and serve a flash application from it that allows a user to record audio from MY hardware (basically record the output of a mic hooked up to the "server") and then download the resulting audio file to his or her computer. Is this possible to do using a single flash application? I'm sure I could send commands to a C application on the "server" that would do the recording, but I'm wondering if there is a more simple solution. Also, monitoring the audio while it is being recorded would be nice.
Thanks

The issue I see with iTalk and similar iPod apps is that the iPod has no line-level input, so you'd have to use one of the earbud/microphone combinations or an external microphone like this one which will be limited as to quality. There are some other options here. But if you want pro quality, and particularly want to tap straight off of a mixing console, a device like the Alesis is going to be necessary, though there may be something less expensive that I can't find.
If a microphone works for you, then iTalk might be fine. There's a freeware version of iTalk, iTalk Lite, you can try, to see if it will do the job. It doesn't have the advanced playback controls and has a limit of 2MB per recording, but it should allow you to test it out. Voxie is another possibility.
Regards.

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.
First, a few definitions:
Monitoring: listening to existing audio while simultaneously recording new audio.
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:
(Sample per second / 1000) * Desired Latency
So, if we are recording at 44,100 Hz and we are aiming for 20ms latency:   44100 / 1000 * 20 = 882 samples. Most audio devices do not allow arbitrary buffer sizes but offer an array of choices, so we would select the closest option. The device I'm using right now offers 512 and 1024 samples as the closest available buffer sizes, so I would select 512 first and see how this performs. If my session has a lot of tracks and/or several effects, I might need to bump this up to 1024 if I experience dropouts.
Now that we hopefully have a pretty firm understanding of what constitutes latency and under what circumstances it is undesirable, let's take a look at how we can reduce it for our needs. You may find that you continue to experience dropouts at a buffer size of 1024 but that raising it to larger options introduces too much latency for your needs. So we need to determine what we can do to reduce our overhead in order to have quality playback and recording at this buffer size.
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.
Device and Driver Options: Different devices may have wildly different performance at the same buffer size and with the same session. Audio devices designed primarily for gaming are less likely to perform well at low buffer sizes as those designed for music production, for example. Even if the hardware performs the same, the driver mode may be a source of latency. ASIO is almost always faster than MME, though many device manufacturers do not supply an ASIO driver. The use of third-party, device-agnostic drivers, such as ASIO4ALL (www.asio4all.com) allow you to wrap an MME-only device inside a faux-ASIO shell. The audio application believes it's speaking to an ASIO driver, and ASIO4ALL has been streamlined to work more quickly with the MME device, or even to allow you to use different inputs and outputs on separate devices which ASIO would otherwise prevent.
We also now see more USB microphone devices which are input-only audio devices that generally use a generic Windows driver and, with a few exceptions, rarely offer native ASIO support. USB microphones generally require a higher buffer size as they are primarily designed for recording in cases where monitoring is unimportant. When attempting to record via a USB microphone and monitor via a separate audio device, you're more likely to run into issues where the two devices are not synchronized or drift apart after some time. (The ugly secret of many device manufacturers is that they rarely operate at EXACTLY the sample rate specified. The difference between 44,100 and 44,118 Hz is negligible when listening to audio, but when trying to precisely synchronize to a track recorded AT 44,100, the difference adds up over time and what sounded in sync for the first minute will be wildly off-beat several minutes later.) You are almost always going to have better sync and performance with a standard microphone connected to the same device you're using for playback, and for serious recording, this is the best practice. 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. Attempt to match the buffer sizes and sample rates as closely as possible, and consider using a higher buffer size and correcting the latency post-recording. (One method of doing this is to have a click or clap at the beginning of your session and make sure this is recorded by your USB microphone. After you finish your recording, you can visually line up the click in the recorded track with the click in the original track by moving your clip backwards in the timeline. This is not the most efficient method, but this alignment is the reason you see the clapboards in behind-the-scenes filmmaking footage.)
Other Hardware: Other hardware in your computer plays a role in the ability to feed or store audio data quickly. CPUs are so fast, and with multiple cores, capable of spreading the load so often the bottleneck for good performance - especially at high sample rates - tends to be your hard drive or storage media. It is highly recommended that you configure your temporary files location, and session/recording location, to a physical drive that is NOT the same as you have your operating system installed. Audition and other DAWs have absolutely no control over what Windows or OS X may decide to do at any given time and if your antivirus software or system file indexer decides it's time to start churning away at your hard drive at the same time that you're recording your magnum opus, you raise the likelihood of losing some of that performance. (In fact, it's a good idea to disable all non-essential applications and internet connections while recording to reduce the likelihood of external interference.) If you're going to be recording multiple tracks at once, it's a good idea to purchase the fastest hard drive your budget allows. Most cheap drives spin around 5400 rpm, which is fine for general use cases but does not allow for the fast read, write, and seek operations the drive needs to do when recording and playing back from multiple files simultaneously. 7200 RPM drives perform much better, and even faster options are available. While fragmentation is less of a problem on OS X systems, you'll want to frequently defragment your drive on Windows frequently - this process realigns all the blocks of your files so they're grouped together. As you write and delete files, pieces of each tend to get placed in the first location that has room. This ends up creating lots of gaps or splitting files up all over the disk. The act of reading or writing to these spread out areas cause the operation to take significantly longer than it needs to and can contribute to glitches in playback or loss of data when recording.

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.

Setting Stereo Mix to default recording device on Windows 8 HP Envy Ultrabook Kills Speakers

When I try setting Stero Mix to be my default recording device, the audio playback always cuts off after about 5-10 seconds or so. When it cuts off it doesn't sound good - it sounds like my speakers fry themselves to a quick and painful death if you can imagine that sound (ok, maybe i'm exagerrating).
Sometimes I have to reboot in order to hear any audio again on this ultrabook. Other times the audio will come back on its own after some time (it may always comes back over time, but most times i'm impatient so I reboot).
Is anyone else with my config able to set Stereo Mix to be the default recordgin and still retain a workable playback device? Anything else I can try?
I have the HP Envy Ultrabook laptop with Windows 8 Pro 64 bit.
Thanks,

I have since discovered stereo mix works when the laptop is plugged in. If it's unplugged it still will not work even though the setting "continue using while on battery power" is checked in the properties. If this is by-design I suppose I can live with this.

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.

No audio output device is installed - Satellite C855D-S5950 - Windows 8 64-bit

Toshiba laptop
Model: Satellite C855D-S5950
P/N: PSCBQU-03G03E
OS: Windows 8 64bit
RAM: 4GB.
Hello,
I have my friend’s Toshiba Satellite C855D-S5950 with Windows 8 (64bit) that about a year old. Some months ago the sound stopped working. He asked me to take a look at it. I have researched this issue and found others have encountered his problem as well.
Currently, the sound icon has a red x, “No audio output device is installed.”
Things I have tried so far to try to fix this:
I went to the Device Manager > Sound, video and game controllers. I have "Realtek High Definition Audio" listed with the little yellow caution icons on them. I have tried to disable, then enable it. I have tried to uninstall the sound driver from the Device Manager. I would restart the laptop, and let the driver reinstall itself, to no effect. I have tried downloading the latest drivers from the Toshiba website. I have tried un-installing the driver while in safe mode from the Device Manager. As well as from Programs and Features. There were no positive change to the audio though, still the same message. The Realtek driver won't finish installing and throws the error: "Install Realtek HD Audio Driver Failure!! [Error Code: 0x000000FF]"
I have gone to Control Panel > Sound, under both the Playback and Recording tabs, it says "No audio devices are installed."
My friend was previously unaware of how to restart or shut down his Windows 8 laptop (as was I until I started investigating this problem for him). I mention this because there are no useful System Restore points to go back to for him.
I have also tried Using the Toshiba Application Installer as well to re-install the driver after un-installing the Realtek Audio driver from Add/Remove Programs. I went to Desktop Assist >Support Recovery > Toshiba Application Installer.
Realtek Audio Driver 6.0.1.6743
It loads most of the way then I get:
"Install Realtek HD Audio Driver Failure!! [Error Code: 0x000000FF]"
I have also tried the following links for additional troubleshooting:
http://forums.toshiba.com/t5/Audio-Sound-Knowledge-Base/Windows-8-Troubleshoot-no-sound-with-Windows...
http://forums.toshiba.com/t5/Audio-Sound-Knowledge-Base/No-sound-from-speakers-or-headphones/ta-p/27...
http://windows.microsoft.com/en-us/windows/no-sound-help#no-sound=windows-7&v1h=win8tab1&v2h=win7tab...
http://forums.toshiba.com/t5/Audio-Sound/No-Sound-quot-No-Audio-Output-Device-is-installed-quot-Tosh...
It was also suggested in the above thread I remove the Microsoft UAA Driver, but could not find this driver on my friend’s system.
Plugging in headphones, toggling the speaker volume controls on the keyboard has had no effect.
My friend has also misplaced his install discs, so I am looking for any solution which does not require the install discs.
Any help would greatly be appreciated.
Toshiba laptop
Model: Satellite C855D-S5950
P/N: PSCBQU-03G03E
OS: Windows 8 64bit
RAM: 4GB.
Thanks in advance.
-Tre

Recovery media is created from the hard drive. You can make it using Toshiba Recovery Media Creator. You can also perform a full system reset to factory defaults using the information on the hard drive without using external recovery media. That sounds like the best course of action now. To do so, use Windows+C > Settings > Power > SHIFT + Restart > Troubleshoot > Reset your PC.
- Peter

Media Encoder not finding "audio capture device"

Hi,
Im using Adobe Flash Media Encoder with Dxtory for Twitch streaming, and at first it worked fine. Recorded both my mic and game audio perfectly. Then seemingly suddenly, I fired it up and it wouldn't stream the game audio. It would my Mic, but thats it. I then unplugged the mic and it started saying I had no Audio Capture Device. I checked in Dxtory and its set up the same as it was when it was working. It picks up both the mic and speakers like it should. I have uninstalled this program multiple times, both 3.1 and 3.2 versions, and each time when I re-download it, it loads up the same profile I had before. So when I am uninstalling it, its not uninstalling everything for some reason. Even though the program is no where to be found after its uninstalled. I have downloaded xsplit, and it picks up the game audio and my mic. So its just this program. Anyone have any ideas? I cant seem to find anything online about this issue.
Thanks!

From where? PP sequence? PP bin? PP Source Monitor? Watch folder?

No "Enhancements" Tab in Windows 8, Recording Device Properties

Hi, I just bought my HP laptop, a HP ENVY 15-q008TX, Windows 8.1. Whenever I'm using the microphone, I've found that the microphone automatically lowers its volume when it's too loud. It's obvious there's some auto-leveling going on in the background, be it by my audio driver or Windows. I don't want that and it's problematic when I'm recording audio for work.
Searching for answers on the internet gave me these common answers:
1) Turn off auto-audio adjustment in Skype -- Done. However, the auto-leveling issue I'm facing happens in other software, even when I'm not using Skype. So it's not a Skype issue.
2) Go to Control Panel > Sound > Select "Do nothing" for communications activity. Done, but issue persists.
3) Go to Control Panel > Sound > Recording Devices > Properties > Advanced > Unselect "Allow applications to take exclusive control of this device". Done, but issue persists.
4) The last solution I've found is to go to the "Enhancements" tab of Speakers Properties AND Microphone Properties and check "Disable all sound effects". This sounds like a possible solution except that my Windows 8.1 has no Enhancements tab at all for both Speaker and Microphone Properties.
I then tried checking the Beats Audio Control Panel (that's the audio software my laptop came with) which has barely any options.
In the "Listening Experience" tab, it doesn't show any auto-leveling option.
In the "Recording Experience" tab, it asks me to attach audio device when my headset microphone is clearly attached and it's working and detected by my Windows and other software.
So at this point I don't know what to do. I suspect there's some audio software issue going on. Any advice is appreciated.

I do not have any issue connecting a microphone. That connecting error in the screenshot seems to be something wrong with the Beats software itself than any issue between my microphone and the HP laptop.
To clarify:
1) My microphone works perfectly and it shows up in Windows Control Panel.
2) I have used my microphone over Skype with no issues.
3) I have recorded many audio recordings with my microphone with no issues.
Where I do have an issue with the microphone, as I have outlined in my original ticket, is that there seems to be an autoleveling or loudness equalization going on where any input audio that goes beyond a certain volume will automatically be softened for 10 seconds, before the microphone increases volume again. I have found this issue reported many times in other forums and apparently it's a common issue, and the solution seems to be to disable the loudness equalization in my Speaker/Mic properties in my audio control panel.
The issue here is that I cannot find this option to disable loudness equalization at all in my audio control panel, be it in Windows or in Beats audio panel (as I have cited in my original ticket). I would like some help to know where exactly I can find and disable the loudness equalization option in my computer because it has to be somewhere since it has been operating in the background.
The manufacturer is Sennheiser.

Audio input device missing since repairing permissions?

Help! I advised my husband to repair permissions on his macbook today so he did and now he's angry with me because one of his audio input devices is no longer showing up in system preferences. It's something called a "firebox" and he uses it to record live audio. He swears it is plugged in correctly and it worked yesterday so now apparently it is all my fault that it doesn't work tonight. He has already tried restarting (my favorite troubleshooting option). What can we do? applecare is already closed, of course, so we can't call.

I've been googling on "Pavillion 23 touch screen". I don't see that model sporting any "line in" port.
Are you sure that's a line in input or the headphone input?
Try to connect a headphone to it and see if sound come out of it.
Say "Thanks" by clicking the Kudos Star in the post that helped you.
Please mark the post that solves your problem as "Accepted Solution"
(Although I am employed by HP, I am speaking for myself and not for HP)

Digital audio recording

Hi,
I will be doing some audio-only interviews which I will need backed-up and then transcribed. I won't be using this audio for an actual video piece, but don't want lousy sound. Does anybody have any suggesions about what type of recording device I should use? I heard that Mini-Disk recorders have the best sound, but I don't know how to get the files onto my computer. And wouldn't I need to buy MANY disks which would add up $-wise? What about MP3 recorders? How's the sound? Do any interface with Macs? And how would I break up any LONG tracks (over 80 minutes) to move onto CDs?
Thanks for your advice!
Eve

Last summer I did a bunch of "man-on-the-street" interviews for a documentary I was shooting. I borrowed an hhb minidisc recorder and plugged a very high-quality Sennheiser broadcast hand-held mic into it.
The whole thing was about the size of a standard-size hardcover book. Only weighed a couple of pounds, too.
I was stunned at the excellent quality of the results. This rig plugged right into the Firewire port on my Mac, and I just dragged the audio files over via Finder copy on to my working hard drive.
The audio files opened right up in my audio editor app. i was able to up-rez them, EQ and tweak them, and then bring them right into FCP. It was very straight forward.
For some of the interviews that were done with video as well as audio, I recorded the audio on the minidisc and wild sound on the camcorder. For sync we just used the time-honored method of shooting a clapboard slate at the from of each segment. Worked great!
I guess this is what radio news guys are using these days for location recording.
However, I agree with the previous poster who emphasized that using a good mic, and putting it in the correct position relative to the subject, is probably 70% of the solution. Regardless of what medium you use to record the audio. Proper technique is always almost everything.
Good luck.

Other ways of adding audio recording to presentations Adobe Presenter 8?

Hi there,
I just realised that I asked a question in the wrong language! I have a quick question re. audio recording in AP8. Is there anyway to add audio to a presnetation other than with a traditional recording device? i.e. can audio be recorded via PC or telepohne and then be added to the presentation? if so, how?
Many thanks!
KH

You can use the Import Audio tool to import any MP3 or WAV audio file. Presenter doesn't care where the audio file comes from, so as long as you can record the audio as one of those two file types, or convert it to one of those file types, you should be able to bring it into Presenter.
To simplify the process, you may want to break up the audio files so there is one for each slide, but this can be done in the audio editor if needed.

T61 audio recording device

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