Sound Studio Frequently Asked Questions
Q: Why do I get no sound when I play an audio file?
A: Sometimes the sound output setup will default to the wrong device. Go to "Audio->Sound Input/Output Setup" in Sound Studio. For Mac OS X, the output section should have "Mac OS X Audio HAL" selected. Go to System Preferences and go to the Sound panel. In the Sound panel, click on the Output tab and select the appropriate sound output device. After this, you may need to close your file in Sound Studio and reopen it for the settings to take effect.
Under Mac OS X, Sound Studio's "Mac OS X Audio HAL (System)" input option corresponds to the selection in the System Preferences->Sound panel's Alerts tab, while "Mac OS X Audio HAL" corresponds to the Output tab. This allows you to send system alert sounds to one device while sending the audio playback to another.
If you're using Mac OS version 9 or earlier, the "Sound Input/Output Setup" dialog shows a list of audio output devices, and you can directly select which device to use. This setting is independent of the setting in the Sound control panel.
If your sound output settings are correct, try quitting and relaunching Sound Studio. If that doesn't work, try restarting your computer.
Q: When I try to record, I get a "Sound quality not available" error alert. What sound quality settings should I be using?
A: Most devices will support 44.1kHz, 16-bit, stereo recording. Try that setting first. If that doesn't work, try 44.1kHz, 16-bit, mono. For USB monophonic microphones, it's important to use the "mono" setting instead of stereo. In almost all cases, you should keep the settings at 44.1kHz, 16-bit, because typically sound hardware will support those settings.
You can also record to a stereo file from a mono microphone, and vice versa. A stereo source or file has two channels of audio, while a mono source or file has one channel. The software can convert between one- and two-channel audio on-the-fly when recording. This means that you can set up your file as a stereo file and record to it from a mono source.
If your input source has 4 channels, Sound Studio might not be able to record from that source, because the app is designed to handle only one or two channels.
Q: The input levels stay in the RED and the waveform is clipped. What can I do?
A: The problem is that the audio signal level, at the point where it is converted from an analog to a digital signal (A/D converter), is too high. This causing clipping, which is a form of distortion.
Software Solution
First, try the input gain sliders in the "Input Levels" window. If you can move them, and if they have an effect on the input levels, you can drag the sliders to the left to reduce the input levels. If the sliders are not active, or moving them has no effect on the input levels, you need a hardware solution.
Hardware Solution
If you've determined that you can't adjust the input gain in your hardware from Sound Studio, check for any other way to adjust the gain. If your hardware includes a input level slider, use that. The Roland (Edirol) UA-30 and UA-3 have such sliders. Use them, even if the level meters on the hardware go into the red. What matters is the level meter in Sound Studio. As long as it doesn't clip in Sound Studio, you should not have any clipping.
What do you do if your hardware doesn't have a input level slider? You need to find another way to reduce the signal level at the inputs. The easy way is to reduce the volume on your source, be it a tape deck, amplifier, or instrument. Most things which output audio have an adjustable output somewhere, usually called a "headphone" jack. You can use the headphone jack, with the appropriate cables, instead of line-level outputs.
What if you don't have a headphone jack? This is where you may need another piece of hardware. You have many options here.
The least expensive is to get a headphone volume control. This is a little headphone extension cord with a thumb-wheel or a slider built-in which will adjust the signal level going through it. One such piece of hardware is the "Koss VC20 Inline Headphone Control."
If you want to spend a little more money, you can get a small stereo mixer which has knobs or sliders for adjusting levels. Then you just put the mixer between your source and your input device. Mixers are multi-function devices, and you can use them to record from microphones and electric guitars, to mix multiple sources, and to send the output to an external signal processor.
Another option is to get an amplifier with headphone jacks or other adjustable output. You could also build your own adjustable attenuator out of electronics parts, but this requires some expertise with electronics.
The thing to remember is that you want to adjust the levels before it goes into the A/D converter (audio input device), because that's where the clipping can occur. After it is turned into a digital stream of numbers, there is nothing you can do to get rid of the clipping, so you need to adjust the levels before that happens.
Q: Why are the input gain controls grayed out and immobile in the Level Meter window?
A: This means that the current sound input device you have selected in Sound Input/Output Setup doesn't support the input gain controls, and therefore you can't change the input gain of that device from Sound Studio.
See "Q: The input levels stay in the RED and the waveform is clipped. What can I do?"
Q: Does Sound Studio support multitracking? Will there be a version which supports multiple tracks?
A: I'm looking into the possibility of adding support for multiple tracks. The current version does not support multiple tracks. It can only record a single stereo track of audio per file. You can record to one file while playing another, but there is no way to synchronize the two files.
Q: How do I record the audio I hear being played back on my computer, such as an Internet streaming audio broadcast or a MIDI file?
A: A basic concept that should be understood is that the audio input channels (what Sound Studio is recording) and the audio output channels (what you hear being played back) are two completely different and independent channels, and that what you hear isn't necessarily what will show up in the audio input channels. In fact, if the audio output were to always appear on the audio inputs, and you had play-through enabled, you would get unwanted feedback in the form of an echo effect or a screeching sound, as if you were to put the microphone in front of an amplifier and loudspeaker.
What you want to do is to physically connect the audio output to the audio input ports. What I mean by ports are the actual 1/8-inch, 1/4-inch, or RCA jacks that physically accept a plug and audio cable. You need to connect the output to the input to intentionally create a loop. If you think of the audio output as pushing audio data out its port, and the audio input as pulling data in, putting a cable between these two will create a loop that lets you record what is being played back. When you do this, you will want to make sure that both software and hardware play-through is turned off, if available in Sound Studio.
This will let you capture any sound being generated by the computer, from software such as Apple Speech synthesis, MIDI data played using QuickTime musical instruments, and streaming audio from RealPlayer or from iTunes, without using any software hacks.
While this configuration will result in some degradation of the signal due to the digital-to-analog and analog-to-digital conversion, the amount of noise is insignificant for most purposes.
To be able to hear what you're recording, you can use a stereo Y-connector to connect the two sound jacks together along with external speakers.
If your computer doesn't have a sound input jack, see you will need an extra piece of hardware called a sound input device or an audio capture interface, such as a USB audio adapter such as the Griffin iMic.
Q: When I plug my microphone into the line-level input jacks, the recorded signals are too low. How can I raise the levels?
A: You need to use a pre-amp to bring the signal levels up to line level. The microphone outputs an electrical signal of only a few millivolts, but line-level inputs are generally around 2 volts, so you need something to amplify the signal. A microphone pre-amp, which can be a stand-alone unit or a part of another device, will properly amplify this signal.
A typical dynamic microphone is made up of a diaphram connected to metal rod inside a coil and magnet set. This device takes vibrations in the air and generates electricity through the coil. The electric signal is very weak, and it needs to be amplified to bring it up to what is called line-level.
Some Macs have audio input jacks which accept microphone-level signals. On those Macs, you can use the input levels gain slider in Sound Studio to amplify the signal. But for most people, your Mac only accepts line-level audio inputs, so you will need a pre-amp. Typically, you would use the pre-amp built into another device, such as an audio mixer board or a tape recorder.
Q: Is there a limit to the length of audio that Sound Studio can record?
A: The absolute limit for the total length of any file in Sound Studio is 2 GB. That's 2 gigabytes, or 2048 megabytes. This limit is for the total length of the file, so if you start with an empty file, you can record 2 GB of audio data. If you are adding to an existing file, you won't be able record as much. Also the file header takes up some of this (usually 54 bytes), but this is small enough to be negligible.
To get the duration that 2 GB represents, you need to know the sound quality settings of the file. This is the sample rate, sample size, and number of channels. The formula for computing the number of bytes given the time in seconds is:
Bytes = Time * Sample Rate * Sample Size * Number of ChannelsWhere sample rate is in Hertz and sample size is in bytes instead of bits. This means that 8-bits equals 1 byte and 16-bits is 2 bytes. To get the time, you would solve for time instead of bytes, so:
Time = Bytes / (Sample Rate * Sample Size * Number of Channels)For 2 GB at 44.1 kHz, 16-bit, stereo sound, this is:
Time = (2 * 1024 * 1024 * 1024) / (44100 * 2 * 2)This result is in seconds, so if you want to convert to minutes, you divide by 60, and if you want to convert to hours, you divide by 3600.
Time = 2,097,152 / 176,400
Time = 12,173.94 seconds
Time = 12,173.94 seconds / 60 seconds per minute = 202.90 minutesTo see how to convert from time to bytes, see "How do I compute how many bytes a recording will take up?"Time = 12,173.94 seconds / 3600 seconds per hour = 3.38 hours
Q: How do I compute how many bytes a recording will take up?
A: To calculate the number of bytes that a recording will take up, given the duration of the recording, you can use a simple equation. First, you need to convert the time to seconds. To convert minutes to seconds, multiply by 60. To convert hours to seconds, multiply by 3600.
30 minutes = 30 * 60 = 1,800 secondsOnce you have the number of seconds, you also need to know the sound quality settings of the file. This is the sample rate, sample size, and number of channels. The formula for computing the number of bytes given the time in seconds is:
2 hours = 2 * 3600 = 7,200 seconds
Bytes = Time * Sample Rate * Sample Size * Number of ChannelsWhere sample rate is in Hertz and sample size is in bytes instead of bits. This means that 8-bits equals 1 byte and 16-bits is 2 bytes. For 1 hour at 44.1 kHz, 16-bit, stereo sound, this is:
Bytes = Time * Sample Rate * Sample Size * Number of ChannelsSince the number of bytes is often such a large number, it is easier to express it in terms of kilobytes, megabytes or gigabytes. In the computer world, a kilobyte is 1024 bytes, instead of 1000 bytes. Also, a megabyte is 1024 kilobytes, and a gigabyte is 1024 megabytes. So to convert the number of bytes, divide by the appropriate factor:
Bytes = 3,600 * 44,100 * 2 * 2
Bytes = 637,040,000
Kilobytes = Bytes / 1024For the above example, the conversions are:
Megabytes = Bytes / (1024 * 1024)
Gigabytes = Bytes / (1024 * 1024 * 1024)
Kilobytes = 637,040,000 / 1024 = 620,156.2 KBTo see how to convert from bytes to time, see "Is there a limit to the length of audio that Sound Studio can record?"
Megabytes = 637,040,000 / (1024 * 1024) = 605.62 MB
Gigabytes = 637,040,000 / (1024 * 1024 * 1024) = 0.59 GB
Q: How do I record from a mono source into a stereo file?
A: There are many different combinations of audio equipment out there which makes it difficult to prescribe a single solution. When people want to record from a mono source, this usually means that their audio source is connected by a single 1/4-inch plug, usually a guitar, microphone, or some other device. What I suggest is that you use a mixer board, pan the signal to the center, and record the mixer's stereo outputs.
If you don't have a mixer board, it becomes more complicated. You could do the mono-to-stereo conversion entirely with pieces of hardware, such as adapters and splitters. Or you could record the audio in stereo, and select one channel of audio and paste it over another in the Sound Studio window. This can take some time as Sound Studio needs to copy every single sample of the file.
Q: I get an error -54.
A: Error number -54 means "permission error," which in the case of sound input drivers means that the sound input is being used by another application. Try quitting any other apps, and turning off Speech recognition or any other software which may be using the sound input. Also, try selecting another device in "Sound Input/Output Setup."
Q: I get an error -229 when pressing record.
A: The error number -229 seems to indicate that your audio input device was unplugged or otherwise made unavailable. Check your cables, and go to "Sound Input/Output Setup" and select your input device again.
Q: Waveform wraps around instead of clips when input gain is set to max.
A: There is a new feature in the Mac OS which emulates an adjustable input gain on hardware that doesn't support such a feature. What is happening is that the audio is being amplified in the software by the OS, after the analog-to-digital conversion. If you set the input gain to the max, low level sounds are increased. A side effect of this is that high level sounds seem to wrap-around in the waveform view.
Another effect is that if you set the gain to a low value, the input audio will start clipping before it reaches the maximum amplitude in the window. What is happening is the analog input audio is being digitized by an A/D converter which has a fixed range of min and max voltage. When the input audio signal exceeds this range, the A/D converter is no longer able to give a meaningful value and returns the min or max value that it can return. This is called clipping. What is happening when you set the input gain to a value other than 1.0 (which is the midpoint on the slider) is that all the input samples are digitally multiplied by the gain factor. Unfortunately, this operation cannot restore the data lost from clipping.
When the gain is set above 1.0, the same multiplication happens, but you get the problem of overflow, because the range of values to represent a sample, typically 16-bits, is limited. When you multiply a sample, and the result is above or below the range of a 16-bit value, the result is wrapped around. When you look at the waveform window, the waveform that goes off the top of the view will seem to wrap around from the bottom.
In this case, it would be best to set the input gain slider to its default, middle position by double-clicking on the slider. This way, the gain factor will be 1.0, and your hardware's input range will be exactly the same as the app's sample data range.
To properly adjust the audio levels before the audio is digitized, see the "Hardware Solution" in the answer to "Q: The input levels stay in the RED and the waveform is clipped. What can I do?".
Q: How do I adjust the left or right input gain independently? How do I adjust the balance while recording?
A: To adjust the left or right channel of the input gain independently, hold down the option key while dragging one of the input gain sliders. Normally, the sliders are linked together, but holding down the option key lets you adjust one or the other independently.
Q: I get a delay when I play an instrument through USB audio. How do I reduce the delay?
A: If you are using software playthrough, you can change the playthrough latency, by following these steps:
To avoid this delay, you need to either use hardware playthrough, which is only available on hardware which explicitly supports this, or avoid using playthrough altogether, by using an audio mixer or other audio hardware to route your instrument's signals directly to your speakers or headphones.
Q: Step-by-step instructions on how to record from a cassette.
A: How to record from a cassette tape player. This assumes that you have already connected the tape player to your Mac using the appropriate cables and adapters, including a USB audio capture device, if necessary.
If after going through those steps you don't have a waveform in your window, try playing back the audio. If you can hear the audio when playing the file and the green playback cue moves, and the time readouts are correct, there is something wrong with the app. Save the file, quit the app, and relaunch it. If it doesn't play back, and the time readouts say 00.000 for the total length of the file, then something went wrong during recording. In this case you will need to go to troubleshooting.
Q: I am monitoring my recording through USB audio on my headphones. How do I minimize play-through delay?
A: A certain amount of delay will always be there if you go through USB, but you can minimize it by going first to "Sound Input/Output Setup" under the audio menu, and then clicking on the "Advanced" button. Then move the slider for play-through latency to the left, to minimize the delay.
One alternative is to avoid going through USB and the computer by reconfiguring your cables so that the audio goes to your headphones before or at the same time it goes into the computer. To do this, you can use a mixing board ("mixer"), or a set of adapter plugs. By plugging your headphones into the mixer, you will hear the audio before it goes through the computer, instead of after. The exact set of adapter plugs you will need will depend on your hardware.
Q: I am trying to record audio that is playing back on my computer. I can hear it on my headphones/speakers, but trying to record in Sound Studio doesn't work.
A: You need to understand that the audio input channels and the audio output channels are two completely different and independent channels, and that what you hear isn't necessarily what will show up in the audio input channels. In fact, if the audio output were to always appear on the audio inputs, and you had play-through enabled, you would get unwanted feedback in the form of an echo effect or a screeching sound, as if you were to put the microphone in front of an amplifier and loundspeaker.
What you want to do is to physically connect the audio output to the audio input ports. What I mean by ports are the actual 1/8-inch, 1/4-inch, or RCA jacks that physically accept a plug and audio cable. You need to connect the output to the input to intentionally create a loop. If you think of the audio output as pushing audio data out its port, and the audio input as pulling data in, putting a cable between these two will create a loop that lets you record what is being played back.
For more information, see Q: How do I record the audio I hear being played back on my computer, such as an Internet streaming audio broadcast or a MIDI file?
Q: How do I set up timer recording (like a VCR)?
A: To set up timer recording, go to the "Automatic Recording" dialog box under the Audio menu. To use it, first create a new file with the appropriate sound quality settings for your recording, and then set up the automatic recording. In this dialog, you can set up the start time and duration or stop time. When you click OK, the file will be ready to start recording at the specified time. You can only set up one recording at a time, and you must leave Sound Studio open while waiting for the recording to start and until it stops.
Q: What is the difference between hard and soft play-thru?
A: Hard is short for "hardware play-through" which means that the hardware is handling the play-through of audio. Soft is short for "software play-through" which means that the application (Sound Studio) is handling the play-through.
The hardware version is preferrable since it introduces no latency. The software version has latency which can be controlled from the Advanced Sound I/O dialog, but which cannot be eliminated completely
Q: I am recording through a MacAlly iVoice or Griffin iMic USB audio device. The input levels seem to go red zone. Is this a problem and what can I do?
A: This isn't always a problem, but you may need to reconfigure your setup or add some hardware.
With the iVoice and the iMic, you won't be able to adjust the input levels from within Sound Studio. As far as I know, the hardware doesn't support changing the input gain. So if you need to adjust the input gain, you will need to find a way to adjust the audio signal level where it enters the USB audio input device.
The first thing to check is the levels in Sound Studio. If you are used to using peak-level meters in the analog world, you'll need to adjust to some changes in the digital world. In the analog world, the first red bar always corresponds to zero dB, and you have some headroom above that where an occasional peak can go. In the digital world, the first red bar is usually somewhat below zero dB (it's at -6 dB in Sound Studio), and you have absolutely no headroom above 0 dB. What happens when you go over 0 dB in the digital world is hard clipping, which is a form of distortion.
This means that as long as the signal doesn't go above 0 dB, your signal will be fine. You could be in the red zone all day, as long as you don't go above 0 dB, you won't have any distortion at all. In Sound Studio, to check that you're not clipping, there are two readouts to the right of the level meter bars. These show the maximum dB level of the input signal, and you can reset them by clicking on them. To use them, click on them to reset them, and then play the loudest passage of your recording through the inputs. If the readouts don't say "Clip", you should have a good recording.
If your input is clipping, you will need to somehow reduce the audio signal level where it enters your USB audio input device. See the answer to "Q: The input levels stay in the RED and the waveform is clipped. What can I do?" for more info.
Q: I have a FireWire DV device (Sony Media Converter, Dazzle Hollywood DV Bridge, etc.) and I can't record audio.
A: You need to have a video signal on the analog video input to in order to record from the audio inputs through FireWire DV. It seems that all FireWire DV devices require this, and I think it is part of the FireWire DV specification. If you don't have the video signal, Sound Studio won't record anything.
To set it up to record through FireWire, you would select "DV Audio" as the device in "Sound Input/Output Setup." Also, make sure you select any device other than "FireWire DV" for the sound output, or you will get an error -14104 when trying to record and playback won't work. It seems you can't have the FireWire DV device play back at the same time it is recording.
This generally means that you will be using the FireWire DV device for input, while using the Mac's built-in audio system for output.
Also, if you have a Dazzle Hollywood DV Bridge, make sure its mode setting is set to "A to D" when recording, and set to "D to A" when playing back audio. When playing back audio, it may take 5 seconds for the device to get ready, so the first five seconds of your playback won't be heard.
If you are able to record only 50 to 60 minutes, try recording at 32 kHz by creating a new file with a 32 kHz sample rate.
Q: How do I connect a phonograph (a.k.a. record player or turntable) to my Mac?
A: To connect your phonograph to your Mac, you will need a phonograph pre-amp to convert the phonograph's output to a line level signal. This is usually built into your stereo receiver or hi-fi amplifier unit, if it has a phonograph input. The purpose of a phonograph pre-amp is to amplify the signal and to apply the standard RIAA equalization curve. If you don't use a pre-amp, the audio will sound bad or you won't hear anything at all.
Connecting the phonograph to the receiver is usually not a problem. See your receiver's instruction manual for that.
To connect your receiver to your Mac, you typically need a "dual RCA to 1/8-inch-stereo" audio cable. Connect the RCA end of this cable to your tape out or tape "record" jack on the receiver. Connect the 1/8-inch-plug end to your audio input hardware, or to the input jack on the Mac.
If you have set up Sound Studio's audio setting properly, you should be able to test the connection. Try playing a record on the phonograph. The input levels window in Sound Studio should show some activity, and you can now record from the phonograph.
Q: General tips for USB audio.
A: If you are using a USB audio device for playback or recording, here are some tips for optimal use:
Q: Is the MOTU 828 supported?
A: No, the MOTU 828 is not supported in Sound Studio. Its driver sends audio data in a format which Sound Studio does not recognize.
Q: I can't control the input gain on my Audiowerk [or Audiophile, etc.] audio card.
A: Your audio capture card probably doesn't support the function which lets Sound Studio control the input gain. There may be a control panel for the card which lets you control the gain. Most add-on audio cards have input gain stages which can't be controlled from within Sound Studio, but which can be controlled from the software that comes with the card.
If you still can't find a way to control the input gain, you may have to use an external means of controlling the levels. See "Q: The input levels stay in the RED and the waveform is clipped. What can I do?"
Q: How do I reduce noise when I'm doing field recordings with microphones?
A: The best thing to do is to prevent the noise from getting into the recording. You should try to move the microphone as close to the subject as possible. If you're doing video recording, get a better microphone than the one built-into the camera. The camera's microphone is often not very good quality compared to separate microphones, and because it's on the camera itself, it can't get close enough to the subject
You have several options for mic'ing your subject. The easiest way is to use a handheld mic which the subject or someone else must hold. The drawback is that for video, the microphone is in view.
You can get a lavalier mic, which clips to the subject's clothing. There are wireless versions of these which lets the subject move around freely. Also, these mics can be small enough to be virtually invisible.
You can also get a boom mic or a shotgun mic, but this requires someone to operate the boom or the mic. If you don't have time to set up a mic but you don't want a microphone in the shot, this would be a better option.
Once you get the microphone close to the subject, you'll find that wind isn't as great an issue as it was before. You will still need to use a wind filter to prevent wind noise from being too loud. Most microphones come with a wind filter or wind sock which you can put over the mic, or you can buy one. But since the microphone is closer to the subject, you won't have to amplify the microphone as much because the sound from the subject is more powerful. (The power of a sound decreases inversely with the distance.) This means that any wind that gets to the mic isn't amplified as much.
Q: I have a 78 rpm record which was played back at 45 rpm. How do I adjust the audio so it is the right speed?
A: You can use the Adjust Pitch command under the Audio menu to correct the speed of the audio. You want to enter a pitch of 173%. This will cause the sample rate to increase. If you originally recorded at 44.1 kHz, it will now read 76.293 kHz. Leave this setting alone for now. Click OK to change the pitch.
You won't be able to play back the file as it is, if the sample rate is above 65 kHz. To fix this, go to the Resample command under the Audio menu, and select 44.1 kHz (CD) for the Rate. Leave the other settings alone. It should start resampling. For a three minute file, it should take about 7 minutes, more or less depending on the speed of your computer.
You now have a file which plays back at the correct speed. However, this isn't the end of it. When you play back a record, a certain set of equalization curves called RIAA curves are applied to the audio. The problem is that the curve for 45 rpm and the curve for 78 rpm are different, so when you play the pitch-shifted audio, it probably won't sound right. While you can fix this in Sound Studio using the Graphic EQ filters, I don't have any suggestion as to what EQ curve to use. There is software out there which will apply the appropriate curves.
Q: Do you have any plans to allow saving a file as an MP3?
A: No, I'm not planning to add the ability to save directly in the MP3 format, because the free iTunes software from Apple will let you convert AIFF files to MP3. Adding such capability to Sound Studio would require licensing the MP3 technology and adding to the cost of a Sound Studio license.
In the future, I might add the ability to load plug-ins into Sound Studio, and a third-party company could write a plug-in that lets you save directly to MP3.
Another possibility is if there are any apps which do MP3 encoding and can accept AppleScript commands, it would be possible in a future version of Sound Studio to have a command which tells that app to encode the file to an MP3.
Q: Can I burn audio CDs in Sound Studio?
A: No, Sound Studio doesn't have the ability to directly burn audio CDs. However, you can use Sound Studio to create the audio tracks that make up a CD, and then use iTunes to create the CD. First, prepare each CD track as a separate file in Sound Studio, and save each file to a folder on your hard disk. Then, in iTunes create a new playlist, and drag your saved files to this playlist. Rearrange the tracks to your liking.
Now, if you have a CD burner connected, you can click on the "Burn CD" button in the upper right corner of the iTunes window. See iTunes Help for more info.
Q: How do I open or edit a track on an audio CD?
A: To open a track on an audio CD, select File->Open, navigate to your Audio CD, and open your CD track directly. The file will be converted into an AIFF file which you can edit and save.
If that doesn't work, you can also use File->Import with QuickTime to open CD tracks.
Q: How do I turn a tape or record into a CD?
A: To make a digital audio CD out of an analog source, you will need several things:
When you have that all set up, launch Sound Studio and go to Audio->Sound Input Source and make sure that Built-In->Sound In is selected if you have built-in sound input, or USB Audio is selected. Please see the USB Audio section if you have trouble here.
Make sure the playthrough option is on, and click OK to close the dialog. If you start your analog source playing, and your Input Levels window is open, you should see the levels meter start working. You want to adjust the levels so that it goes into the yellow occasionally, but not into the red. (See the Input Levels section for the advanced discussion about levels.)
Once you've set up the levels, you can start recording. Select File->New and create a 44.1kHz, 16 bit, stereo file. You can start recording now. The maximum file size is 2 GB, which translates to about 3 hours and 20 minutes at the current settings. The best way to record is to do an entire side of the record or tape in one pass, and then edit later.
When one side is done, click stop. The window will start to redraw, but you can start editing right away. You will want to turn each song or section of the recording into a separate file so that your CD burning software can make them into separate tracks. The fastest way to do this is to place markers at the start of each song, and then use the "Edit->Split By Markers" command. You can name each marker according to the song that comes after it. When you use the "Split By Markers" command, you will need to specify where to store all the files that will be created for each piece of audio separated by markers.
You can now take the files you have just created and drag them into your CD burning software (such as Roxio Toast or iTunes) to make an audio CD. Make sure you set the track gap to zero if you want seamless transitions between tracks.
Optionally, after splitting the side into separate files, you can clean up the ends by deleting extra silence or use fades. Also, if some tracks seem too quiet, you can use the Normalize command to bring the volume up, without the risk of making it too loud and clipping samples. Save each file as an AIFF file, and the CD burner software should take it from here.
Q: WAV files can't be opened in Sound Studio, or WAV files saved by Sound Studio aren't openable on a Windows program.
A: The Windows WAV file format is a complex file format with many programs expecting certain non-standard pieces of data in the file. Sound Studio only supports the standard WAV file format with uncompressed audio sample data. It does not support the various compressed WAV file formats, such as MP3 or IMA inside the WAV file. (You can import compressed files with the "Import with QuickTime" command.)
To open a WAV file in Sound Studio, first make sure it is not compressed. You can check this by using QuickTime Player. Also, make sure that the file's type is "WAVE" or the file name ends with ".wav". If it still doesn't open, try using the "Import with QuickTime" command to open it. If that doesn't work, then your file was probably corrupted in the transfer.
If the WAV file you save in Sound Studio can't be opened on a Windows program, try opening the file with Windows Media Player. If it works in WMP, they your program may have some special requirements for opening WAV files.
Please make sure to include the name of the program which created the file, or the program with which you're trying to open the file, in any bug reports.
Q: How do I save in a compressed file format?
A: To save a file with compression, use the AIFF-C file format. After specifying where to save the file, it will present a compression type dialog. The suggested format is IMA 4:1, which is a lossy compression format which results in files that are one-quarter the size of the uncompressed version.
Alternatively, you can use iTunes to use MP3 compression. See the FAQ entry for iTunes for more info.
Q: How do I find out how many bytes of disk space a file takes up?
A: To find out how big a file is, in terms of bytes, use the Finder. After you save your file in Sound Studio, go to the Finder, select the file you're interested in, and go to "File->Get Info." An "Info" window should pop up. In this window, the "Size" field will indicate how many bytes of disk space the file uses. A "KB" is a kilobyte, which equals 1,024 bytes, and a "MB" is a megabyte, which equals 1,024 kilobytes or 1,048,576 bytes.
Q: How do I edit an MP3 file?
A: To edit an MP3 file, you need to import the file into Sound Studio, do the editing, and then save as AIFF so that iTunes can turn it into an MP3. Here are some detail instructions:
In Sound Studio, select File->Import with QuickTime, and choose the MP3 file you wish to edit. After importing the file, you'll have a new window with the audio data in it.
If you want to bring the volume up to their maximum possible level, you can normalize the audio. Go to Edit->Select All to select all the audio. Then go to Filters->Normalize, and use the default setting of 96%. This will take some time to do. Wait for it to finish normalizing.
Go to File->Save As and save the file as an AIFF file, on your desktop. When it's done saving, create an playlist in iTunes 3 and add the AIFF to the playlist. Then select the file in the playlist and select Advanced->Convert Selection to MP3. The new MP3 file will be placed in your iTunes music folder, in the "Unknown Artist/Unknown Album" directory. You can move this MP3 file where you wish. The ID3 tag information is not saved in this process, so you will have to re-enter them by hand.
If You Don't Have iTunes
If you can't get iTunes, you can try other software that does AIFF-to-MP3 conversion. There are many apps out there that will do that, all of which require paying a fee for the full use of the MP3 encoding function. Some of the apps are:
Q: How do I convert to MP3 with iTunes 3?
A: In Sound Studio, go to File->Save As and save the file as an AIFF file, on your desktop. When it's done saving, create an playlist in iTunes 3 and add the AIFF to the playlist. Then select the file in the playlist and select Advanced->Convert Selection to MP3. The new MP3 file will be placed in your iTunes music folder, in the "Unknown Artist/Unknown Album" directory. You can move this MP3 file where you wish. The ID3 tag information is not saved in this process, so you will have to re-enter them by hand.
Q: How do I remove hum, tape hiss, or other noise in my recordings?
A: Noise is a random phenomenon present to a certain degree in all recordings. Because it is random, there is no easy way to remove it from a recording.
The best situation is to prevent as much noise as possible from getting into the recording. It is very difficult, and can be expensive, to remove noise once it has gotten into the mix. If you can't get another recording or do another take, there are a few things to check before turning to the high-end noise removal strategies.
For tapes, check if it was recorded using Dolby NR or dbx noise reduction. If it was, you should switch on the appropriate Dolby or dbx noise reduction filter or run the signal through the appropriate Dolby or dbx decoder.
In Sound Studio, there are a few things you can try to make the noise less noticeable. You can use the Graphic EQ to reduce the high frequencies, but this will also color the sound and may make it muddier.
If there is still noise, you'll need the expertise of an audio engineer who knows this kind of stuff, or a book about audio recording, because this is a tricky subject. There is software available that can process and reduce noise, such as Arboretum's Ray Gun, and they cost anywhere from a few hundred dollars to a few thousand.
Q: How can I remove clicks, pops, and crackle from vinyl records in Sound Studio?
A: For snaps, crackles, and pops, there are only two good filters. One is the Interpolate filter, which is best used for individual clicks and pops. The other is the Fade Special filter, which is best used to erase any areas where there are too many pops to take out individually.
The Interpolate filter should be used at the 1:1 zoom level. You select the pop plus a few samples before and after it, and then apply the Interpolate filter. The pop looks like a spike in the waveform. The result of the filter should look like a nice rounded waveform. If it doesn't come out right, you can Undo it and try again on a slightly different selection.
The Fade Special filter should be used with a 'U' shaped envelope. The middle of the area you select will be erased, so you should use this only on an area of about 1 to 10 milliseconds.
Q: How do I edit the audio track from an iMovie clip?
A: If you are using iMovie, and want to do more advanced editing of the audio track in a video clip you have in iMovie, you do that with the help of Sound Studio. The first thing to do is to have a video clip and put it into the timeline of the iMovie project. Then use "Advanced->Extract Audio" on the clip in iMovie. Now the video track will have a separate audio track locked to the video clip. "Locked" means that the beginning of the audio is synchronized with the beginning of the video. The extraction may take a few minutes for long clips.
Now, you can find the audio file in the "Media" folder in your iMovie project folder, in the Finder. The file may be named "Voice 01" but the name may be different. You can find the file name by selecting the audio clip in iMovie and going to "File->Get Clip Info," and the file name will be the "Media File."
Once you know which file to edit, you can open the file with Sound Studio and do any kind of editing and apply effects to the audio. The changes won't show up in iMovie until you save the file in Sound Studio.
Q: How do I cross-fade from one file to another?
A: You can cross-fade from one file to another in Sound Studio. First, open two files. Preferably, will want to make sure they both have the same sample rate, sample size, and number of channels. This example uses a cross-fade of five seconds, but you can use a different duration. Apply the Fade Out filter to the last five seconds of the first file. Apply the Fade In filter on the first five seconds of the second file. Now, select all of the second file and select Edit->Copy. This will put it on the clipboard. Use "Mix" (under the Edit menu) to paste the first file at a point five seconds before its end. The audio on the clipboard will be mixed on top of the first file at the insertion point, and the file will be extended to hold the new mix. And there you have your cross-fade.
Q: Can I edit out pops and clicks in Sound Studio?
A: Yes, you can edit them out. You must individually locate each pop and click, zoom in to 1:1, and select the click and a little area around it. Then use the "Filter->Interpolate" filter. This will smooth out the waveform where you have select. You may have to undo it and try it again with a slightly different selection until you get the hang of the filter.
Q: How do I remove a marker?
A: You have two ways to remove a marker. Both require you to first check Show Markers and uncheck Lock Markers. The first method is to drag the marker out of the window. The second method is to double-click on the marker, and click the Delete button in the dialog that shows up.
Q: Adjust Pitch: Can I change the pitch without affecting the duration/tempo?
A: The "Adjust Pitch" command is a tape-style pitch adjustment. This means that when you change the pitch, the tempo and duration of the audio is also affected.
So the answer is no, it can't change the pitch without affecting the duration and tempo. There are programs and hardware units which can do pitch shifting without affecting the tempo. Search the Internet for pitch shifting and slowing down the tempo.
Q: How do I change the key from Gb to F?
A: The difference between Gb and F is one semi-tone, so you would want to adjust the pitch down by one semi-tone. This is about 94.4%, so if you use the Adjust Pitch command and enter 94.4% in the pitch field, it should change the key of the audio down one semi-tone.
If you want to adjust it by more than one semi-tone, you can calculate the percentage by using the formula: percent = 2 ^ (semitones/12).
Q: What can I do if my scratch disk is full?
A: If Sound Studio gives you a message saying that the scratch disk is full, that means that the hard disk where the app stores its working files has no more available, free space, and you need to either delete some files on that disk, or select a different disk with more free space.
First, you will need to determine which disk you're using as a scratch disk. Select "Preferences" from the menu, and click on the "File" tab (it should already be selected). In the "Scratch Disk" pop-up menu, it should say which disk is being used as a scratch disk. By default, this is "Startup" in italics, which means that whichever disk used to boot up your computer and contains your active system software is being used as the scratch disk. If you select the menu, below the "Startup" item is a list of disks on your computer.
To free up space on your disk, first determine which disk is being used as your scratch disk. Then, in the Finder, find out what files and folders are taking up space on your hard disk. Make sure you empty the trash if it has any files in it.
To select a different disk, select it from the pop-up menu. If you only have one disk available, you won't be able to select a different disk. If you have more than one disk, you can find out how much space is available on each disk by using the Finder. Select the disk, and go to "Get Info" in the File menu in the Finder. The "Available" field tells you how much space is free on the selected disk. Find a disk with the most free space, and use it as the scratch disk.
If you still don't have enough space, you can try adding a hard disk to your system. There are both internal and external hard disks, and if you can use the disk in the Finder, you can use it as a scratch disk in Sound Studio.
After selecting a different scratch disk, you will need to save your files and quit Sound Studio. Then relaunch it in order for the scratch disk settings to take effect.
Q: How do I layer two sound files on top of each other?
A: You can mix two sound files together into one sound file. To do so, follow these steps:
The items in parentheses above indicate menu items to be selected.
Q: Why are all the items under the Filter menu disabled?
A: The filters only work on selected audio. You must select some audio in order to apply the filter. Try creating a new window, recording some audio or inserting some noise, and selecting all of it. Then you can use the filters.
Q: Can effects or filters be previewed in real-time, before applying them?
A: There is a "Preview" button for most filters that will let you preview the effect of the filter on the selection. Click on the preview button, and it will run the selected audio through the filters with the settings shown the dialog box.
Skipping Issue
If your audio starts skipping, this means that your computer is not fast enough to process the filter in real-time. (To be fair, certain filters, including the Graphic EQ, are not optimized enough to run in real-time on any currently shipping computers.) This skipping will not be in the final application of the filter when you press OK.
You may wonder why the preview button exists in certain filters if they can't be run in real-time. Most of the filters will run fast enough to run without skipping on all currently shipping Mac computers. The only filters which skip on a 667 MHz G4 are the Graphic EQ, Flanger, and Chorus. The problem with these filters is that the trigonometric functions such as sine are too slow. Some of the other filters may skip on slower machines. Whether a filter can run in real-time on a machine depends on both the hardware and the operating system software. Rather than remove the preview feature from certain filters, I prefer to include it in case someone wants to try it out.
Q: How do I adjust the duration of a file?
A: To change the duration of an entire audio file, select Audio->Adjust Pitch and enter the a new duration for the entire file. This is a tape-style pitch adjustment, so the pitch of the audio will change with the duration. As long as the change is not more than 5%, it shouldn't sound noticeably odd.
After using Adjust Pitch, if you want to put the file on a CD, you need to resample the file to 44.1 kHz. Use "Audio->Resample" and put in 44.1 kHz for the sample rate.
Q: How do I make my audio sound like it came through a telephone?
A: Try using the Low-Pass filter first to remove all frequencies above 3400 Hz, with a window width of 95, then use a High-Pass Filter to remove all frequencies below 400 Hz, with a window width of 255 (you have to type in this number). You can adjust the window width to change the frequency roll-off. The frequency graph in the filter set-up dialog will show you the effect of the filter.
Q: I have several files with widely varying sound levels. How do I get them all to the same level?
A: You need to use the Normalize filter. You can work on each file one at a time. For each file, do this:
Q: I cannot save Presets.
A: Check that your Presets folder path is set to a directory (folder) where you have write access. (To check the Presets path, go to Preferences.) You can set the presets folder to any directory, but you should set it to one where you have permissions to write to the folder if you want to save presets.
Q: How do I add a delay to a single channel of audio?
A: The easy way to introduce a delay is to select the channel you want to modify, and use the "Delay/Echo" filter. The settings you want to use are:
Dry mix: -90 dB
Wet mix: 0 dB
Delay time: variable, depending on how much delay you want
Feedback: off
You have to type in -90 dB for the dry mix. The slider doesn't go any further, but the numerical value will be honored.
Q: Filter preview crashes on a dual-CPU computer.
A: There's a bug in the filter preview which can cause it to crash on a dual-CPU computer. If your Mac has dual processors, and you bring up a filter and click the preview button, there is a chance that Sound Studio will unexpectedly quit with an error. The bug has to do with some data inconsistency when the two processors try to access the same data. The current workaround is to avoid previewing, and instead apply the filter to small section by selecting a small section, clicking OK in the filter dialog to see how it sounds, and then undoing it.
Q: How do I remove vocals from a song?
A: There is equipment and software designed specifically to remove vocals (or "devocal"). Sound Studio does not have any specific filters for doing this, but it may be possible to remove some of the frequencies associated with vocals. Using the Graphic EQ, you can set certain frequency bands to the minimum setting to remove them. You may have to experiment to find the correct frequencies to remove. Another technique is to take the difference between the left and right channels. This technique, by itself, is not very effective, but combined with the use of EQ, can help remove vocals. Both of these methods are somewhat technical, so I suggest searching the web for "devocal" or "remove vocals" for more info.
Q: Is there a version of Sound Studio in French (or German, Italian, etc.)?
A: No, Sound Studio has not been localized for any language other than English yet. I have tried doing German and Italian localizations, but I think it would be better to concentrate on the English version than to have a less-than-excellent localization. I'm open to any suggestions on this topic.