Video capture cards

If you have no way to utilize FireWire, you have to use a video capture card to convert the analog video signal into digital information. Video capture cards, like cameras, are available at a wide range of price points. The more you spend, the higher quality your video capture will be.

Video capture card settings
Using a video capture card involves connecting your camera to the capture card and then specifying the settings for your capture. Where and what settings you can specify depends on your capture card and your video-editing platform. Essentially, you specify the following:

Resolution: The dimensions of your screen capture

Frame rate: How many frames per second to capture

Data rate (or compression): Whether to capture the video uncompressed or use a codec during capture

You also may be able to adjust the video settings for your capture, such as brightness, contrast, and saturation. However, the adjustments offered by most budget capture cards tend to be fairly coarse. A better approach is to digitize your video as purely as possible and to do your adjustments using video-editing software, which enables a much finer degree of control — and the ability to "undo."

To capture the highest quality video possible, try to capture full screen, full frame rate, and uncompressed. You may have to scale back, however, depending on your hardware situation. Full-screen uncompressed capture is a very data intensive process, requiring a fast machine and lots of storage. If you have to scale back, start by trying to use a different encoding scheme such as YUV. If your machine still can't capture full frames reliably, you'll have to reduce your resolution, possibly to 1/2 size (320×240). This is a perfectly acceptable starting point for a video podcast, provided you can capture at full frame rate.

Digitizing via a capture card using SwiftCap: A step-by-step example
If you are capturing video via a capture card, you can choose some settings. Most video-editing platforms come with a video capture application that allows you to access your video capture card and adjust your settings. In this example, we'll use SwiftCap, which is the video capture application that comes with all cards sold by Viewcast, one of the more popular video capture card manufacturers. It has some nifty features that you'll find handy for successful video captures.

Follow these steps to digitize video with SwiftCap:

1. Make sure that your capture card is installed correctly and that your camera or videotape deck is connected to your capture card.

2. Open SwiftCap. Provided your camera is running, you should see a preview of your video (see Figure 1). If you don't see a video preview, make sure "Preview Video" is checked on the View menu.


Figure 2: The ViewCast SwiftCap application

3. If you're still not seeing video, make sure your video source is correct. Many video capture cards have multiple inputs, for example, a composite input and an S-Video input. To check your source setting, choose "Capture Settings" from the Settings menu or click the capture settings icon (see Figure 2).

4. Select the appropriate source from the drop-down Source menu on the left side of the Capture Settings window (see Figure 3).


Figure 3: Select the appropriate source in the Capture Settings window to make the preview active.

5. Before you do any capturing, make sure your system is capable of capturing the screen resolution and encoding scheme you want to use. Ideally, you want to capture full screen (720×486, or 640×480 if your capture card automatically scales the input), but this requires a fast computer and plenty of storage. SwiftCap includes a handy disk performance analyzer that can save you lots of woe. From the Tools menu, choose "Disk Performance." This brings up the Disk Analyzer shown in Figure 4.


Figure 4: Click "Profile Drive" in the Disk Analyzer window to analyze your hard drive performance.

6. Select the drive you intend to capture to in the Local Drives pane, and click the "Profile Drive" button. The analyzer then determines the speed at which you can capture video and displays the results in the lower right. You can see in Figure 4 that this drive is capable of capturing only 20.8 frames per second using RGB 32 encoding. We'll have to choose a different encoding scheme to get the full frame rate.

Tip You should always capture video to a drive other than the system drive if possible.

7. The idea is that you have to get your frame rate comfortably over 30 frames per second. YUV encoding is more efficient, so you can usually get your frame rate up considerably by switching to a YUV encoding scheme. In Figure 5, you can see that switching to YUY2 puts our potential frame rate over 40, which is far more than we need.


Figure 5: Using a YUV encoding scheme is more efficient and allows higher frame rate captures.

8. After you've found settings that work with your system, go back to the Capture Settings window and enter those settings (see Figure 6).


Figure 6: Be sure to choose the right settings in the Capture Settings window.

9. Next, specify which drive you want the captured file saved to. Select "Capture Destination" from the Settings menu, or click the capture settings icon to bring up the Capture Destination window (see Figure 7). Clicking the double arrow button at the top right opens a browse window where you can specify a location and a filename. Be sure to use the same drive that you tested in Step 5.


Figure 7: Specify where to put your captured video file.

10. You should now be ready to capture video. Rewind the tape to just before where you want to start capturing, start tape playback, and then start the capture process by selecting "Start" from the Capture menu or by clicking the start capture icon.

11. SwiftCap disables the video preview during capture, so you have to either monitor via your camera's monitor or an external monitor. When you've captured what you needed, choose "Stop" from the capture menu or click the stop capture icon.

12. After you stop your capture, SwiftCap displays a capture results window. It is critical that your capture have zero dropped frames. If your capture dropped frames, you should recapture your video using a different encoding scheme or a smaller capture resolution.

That should be it. Using a capture card requires a few more steps than using a FireWire setup, but you should be able to get equally good quality, provided you're using a good quality capture card.

FireWire & setting it

By far the easiest method of capturing video is via FireWire. FireWire (officially known as IEEE 1394, and also known as iLink) is a standard by which data can be exchanged at very high speeds using a special cable and a FireWire port. Most digital video (DV) format cameras include a FireWire port. In addition to data transfer, the FireWire standard includes the ability to control remote devices (such as a camera). This makes capturing video from your DV camera a snap.

Simply connect the camera to your FireWire port, and open your video-editing platform. Most video-editing platforms include some sort of built-in "import" or "capture" functionality. You can generally control the camera from within the application and specify what part of the tape you want to capture. If you want to capture the whole tape, just rewind and hit the capture button.

If your computer doesn't have a FireWire port built in, you can buy a FireWire card for less than $50. If your camera doesn't have FireWire built in, you can buy a digital video converter. Digital video converters take analog audio and video inputs, for example from your camera, and create a DV signal that is available on the FireWire port. An added benefit of digital video converters is that they work both ways, meaning you can send the output of your video editor to the FireWire port, and the digital video converter converts it back to an analog signal that can be displayed on a monitor for quality-assurance purposes.

FireWire settings
One of the advantages (some might say disadvantages) of digitizing via FireWire is that there really are no settings for you to adjust. The DV standard is hard-coded into the entire process. You'll always be capturing full screen, full frame rate, using the DV codec.

The DV codec is another advantage to capturing via FireWire. Digital Video (DV) is compressed as it is recorded, making DV files about 1/5 the size of uncompressed video files. This makes them easier to store and move around. The compression affects image quality, however, and is one of the reasons that DV isn't considered true broadcast quality. However, the ease of use and the price points of DV cameras are virtually impossible to argue with.

Capturing via FireWire isn't really capturing in the truest sense; it's really just transferring the file from your camera (or tape deck) to your computer, just like you'd transfer a file from one computer to another. Because it's just a file transfer process, the video on your computer is an exact copy of the information on your camera.

Transferring files via FireWire using iMovie: A step-by-step example
Like so many things in the MacOS world, transferring DV from your camera to your Mac is a snap. Follow these steps:

1. First, connect your camera to your Mac's FireWire port, and turn your camera on to the playback or VCR position.

2. Open iMovie.

3. iMovie automatically detects that you have a camera connected and opens in "camera mode" (see Figure 1).


Figure 1: iMovie automatically detects a camera connected to your Mac and opens in camera mode.

Tip In camera mode, the stop, play, fast forward, and rewind buttons control your camera. You can use these controls to find the portion of your tape that you want to import.

4. To begin the import process, click the Import button (or hit the spacebar). If the tape is already playing, iMovie starts importing video from that point. If the tape is stopped, iMovie starts tape playback and begins importing.

5. iMovie automatically breaks up the imported video into clips each time it senses a new scene. This can be very handy, but it also can be a problem if you're trying to import a pre-edited piece of video with multiple scenes. You can disable this feature. From the iMovie menu, select Preferences (or just hold down the Apple button and push the comma key). Select the Import tab, and then deselect the "Start a new clip at each scene break" option.

That's it! You can now drag your clips to your iMovie timeline and edit away.

How compression works

How compression works
Most compressors offer the same basic controls, which allow you to set the following:

Threshold: Where the compression effect kicks in

Ratio: The amount of compression applied above the threshold

Attack and Release times: The length of time after the threshold is crossed that the effect is applied and removed

Figure 1 illustrates what different compression curves look like. Looking at the curve, you'll see that signal levels below the threshold are unaffected, and signal levels above the threshold are attenuated. The higher the compression ratio is, the more attenuation. When the compression ratio is high, it is known as limiting, because it more or less prevents the audio from exceeding the threshold.


Figure 1: Compression curves with different compression ratios

Setting a threshold
To illustrate how different threshold settings affect the output, let's assume that we're working with the audio file illustrated in Figure 2. We can see that this file has peaks as high as -2dB, but the bulk of the file is below the -10dB mark. If we want to compress this file lightly, we should set a threshold in the -6dB to -10dB range. Figure 3 illustrates compression applied to this file using two different thresholds, -6 and -20dB.


Figure 2: The audio file from Figure 7.5 after compression using a threshold of a) -6dB and b) -20dB


Figure 3: The audio files from Figure 2 after applying compensating gain

What is immediately apparent is that the file in Figure 2b has been compressed far more heavily than Figure 2a. We need to apply some gain to restore these files to their former levels. Figure 2b has far more headroom, so we can apply much more gain. After applying gain, we end up with the files illustrated in Figure 3.

These files are both much louder than the original, but if you look closely at the figure on the right, the entire file is loud. It doesn't have any dynamics left, because the dynamic range has been compressed. To be honest, this file might be a little too compressed. Files that have been over-compressed are fatiguing to listen to, because EVERY SINGLE SYLLABLE IS LOUD. Think of drive-time radio programs; they're highly compressed, because the DJs are going absolutely nuts in the studio. The idea is to compete with all the noise of traffic and to keep you awake on your drive to and from work. But this is not the type of programming you really want to listen to all day long (see the "Compression: How Much Is Enough?" sidebar).

If your original audio file is well recorded, you should have peaks in the -3dB to -6dB range. Choosing a threshold in the -6dB to -10dB range is a safe starting point. This way, you're only compressing the loudest sections of your file, leaving most of your file untouched. If you find yourself dropping your threshold much below that, you may consider revisiting your signal chain to figure out why your recording is so quiet in the first place.

Setting a ratio
The ratio setting determines how much compression is applied over the threshold. For example, a 2:1 compression ratio means that for every 2dB by which the incoming signal exceeds the threshold, only 1dB of gain is applied. Ratios up to around 4:1 are mild and can be used safely, provided you set a sensible threshold. Ratios in the 4:1 to 10:1 range are fairly heavy and should be used with caution. Any ratio over 10:1 falls into a special category known as limiting.

Limiting can be useful as a preventative measure, but it isn't appropriate as your main form of compression. For most applications, start off with a ratio of 4:1 and experiment with using slightly more or less until you achieve the effect you're after. In particular, voices are particularly compression tolerant, so if you don't have any music in your podcast, you may be able to use more compression (see the "Compression: Voice versus Music" sidebar).

Setting attack and release times
The attack and release times control how quickly the compression effect is applied to signals that exceed the threshold you set, and how quickly the signal level is returned to the original input. For most content, you want a quick attack time, so signals that exceed the threshold are immediately attenuated. For the release time, you want something a bit longer, so the sound doesn't abruptly get returned to the original level.

This is fairly self-evident from the attack and release controls. The scale on the attack control knob generally is in milliseconds, and the scale on the release knob is in seconds. Start with a quick attack, say 10-20 milliseconds, and a gradual release around 500 milliseconds. These settings should work for most podcasting content, but don't be afraid to play around with your compressor to see how these settings affect the compression.

Bit depth and sampling rates

Bit depth and sampling rates
Bit depth and sampling rate are the two things that determine the fidelity of a digital audio file. You may have seen pairs of numbers rattled off in equipment brochures such as 16/44.1, or 24/96. The first number generally refers to the bit depth, the second to the sample rate.

In the previous section, we talked about taking measurements of an analog input at specific intervals. This is the sample rate. As the sample rate increases, the digital representation more closely resembles the analog original. You must choose a sample rate that is high enough to give you the fidelity you require.

Bit depth is the other variable in the fidelity equation. Bit depth refers to the number of bits used to represent the measurement of the analog voltage. For example, you could represent the distance to the nearest street corner as a block, 50 yards, 143 feet, or 1,717 inches. As you increase the accuracy of your measurement, you need more digits to represent the number. The same is true for our digital samples of our audio input.

In the binary world, each bit you add doubles the range of numbers you can represent. Looked at in another way, each bit you shave off your bit depth cuts your accuracy in half. If we use eight bits, we can have up to 256 different values. With 16 bits, we get more than 65,000 possible values, and 24 bits provides more than 16 million values. So how much accuracy do we need?

Choosing your digitization settings
Ideally, you want to record using the highest sample rate possible, and use the largest bit depth you can. The problem is that using high sample rates and large bit depths creates larger files. Increasing the bit depth from 16 to 24 bits increases the file size by 50 percent, and increasing the sample rate from 44.1KHz to 96KHz more than doubles the file size. Even though storage is relatively cheap and getting cheaper all the time, there are some practical limits to how much fidelity you really need, especially given that podcasts are encoded using lossy codecs such as MP3 that compromise the fidelity. A better master provides a higher podcast quality, so how much fidelity do you really need?

Compact discs use 16 bits and a 44.1 KHz sample rate. Most people consider this sufficient to capture the full range of audio that the human ear is capable of hearing (though many audio engineers would disagree). This is probably what you should use for your master recordings. Although many recording devices and audio editing platforms now offer higher sample rates and bit depths, it's debatable whether the extra quality justifies the increase in storage requirements. You'll just end up using lots more storage space for your archived masters.

If you're producing your podcasts to an extremely high standard, consider using 24/96 (24 bits/96KHz sample rate). This "future-proofs" your masters and will impress all your audio engineering buddies. For most folks, however, the standard 16/44.1 setting will more than suffice.

Tip If you're producing a video podcast, Digital Video (DV) audio is sampled at 48KHz, or sometimes even at 32KHz. These sampling rates were chosen when digital video standards were developed. They're different from the 44.1KHz audio sampling rate, because the sampling rate for audio predates them and was chosen due to technological restrictions at the time. This is fine; the key is not to resample the audio to 44.1KHz (or any other sampling rate). Re-sampling introduces artifacts and degrades the quality of the original audio.

Basic Audio Production

Now that you've spent time drooling over the latest and greatest audio gear, and invested some of your hard-earned cash in decent equipment, you need to figure out how to hook it all up and produce professional sounding podcasts. The great thing about working with audio is that for a minimal investment, you should be able to produce your podcast to a very high standard. The powerful technological leaps we've seen in the world of computers have also brought great advances (and price drops) in the world of home recording. What once required thousands of dollars worth of equipment now costs hundreds, or less.

We will start off showing you how to connect your equipment to get the best sound, and then talks about some general recording techniques. Audio production may seem daunting at first, but by setting up some simple procedures and sticking to them, you'll find it to be pretty simple, and more important, lots of fun.

After that, we cover editing, where much of the power of audio production actually lies. Good editing can transform your podcast from mundane to professional. The techniques described are all standard operating procedure in radio, television, and recoding studios around the world. Though we can't hope to turn you into an audio engineer in a few short pages, we can at least point you in the right direction. Let's start by setting up your equipment — the right way.

Setting Up Your Equipment
Different kinds of equipment you need to produce your podcast to a high standard. Ideally, you took the plunge and bought some equipment to fit your budget and the scale of your production. Now it's time to unpack everything and connect everything together. This is actually a critical step in podcast production. If you set up your equipment incorrectly, you'll leave yourself vulnerable to noise, interference, and distortion, which will compromise the sound quality of the final production. If set up correctly, your hardware will have you on the road to creating broadcast-quality programming. To understand why this step is important, you have to understand the concept of gain.

Setting your levels
Gain, also known as level, is the measure of the power of your audio signal. When using analog audio equipment, such as microphones and mixing desks, the signal is a continuously varying voltage. The higher the voltage is, the higher the gain and the louder the audio. All audio equipment is designed to work within a certain known range of voltages. To obtain the best possible quality out of your audio equipment, without adding any noise or distortion, you want to work within the optimal range for that piece of equipment, known as its dynamic range.

Dynamic range
The dynamic range of a piece of audio equipment is the difference between the loudest sound it can handle without distortion and the internal noise floor of the equipment. For example, when you turn a portable radio up too loud, you'll hear the sound crackle and buzz; that's distortion. You've just exceeded the dynamic range of the radio. The noise floor lies at the other extreme of the spectrum.

All audio equipment produces some amount of noise; there's no such thing as a perfectly quiet piece of equipment. That's because they're imperfect by definition. Every piece of audio equipment has all kinds of electronic components, each one adding a minute amount of noise, which taken in total is the noise floor. You can hear this noise — just turn your stereo up really loud while you're not playing anything. You'll hear a hissing and possibly a buzzing noise. This is the system noise that is being amplified. If you were actually playing a CD, you wouldn't hear this noise, because the music would be much louder than the noise.

More expensive equipment uses better components, which produce less noise. Consequently better equipment has a greater dynamic range. Cheaper equipment, well, you get the idea. This is the argument for investing in decent audio production equipment. If you produce audio with no audible noise, your podcast sounds much better. Noise is a dead giveaway that an amateur is behind the controls. Another giveaway is distortion. After your signal distorts, you can't remove the distortion. It can't be edited out of the signal, and it compromises the quality of your podcast.

Dynamic range is measured in decibels (dB). The human ear is capable of perceiving up to 120-130dB of dynamic range, before the pain threshold kicks in. We can hear a faucet dripping down the hall in the middle of the night, and endure hours in front of our favorite rock band. Our ears are extremely sensitive, which is not necessarily the case with the equipment and/or transmission methods used to produce audio.

Different audio transmission methods have different dynamic ranges. For example, compact discs have about 96dB of potential dynamic range, whereas FM radio has only about 70dB of dynamic range and AM radio has only about 48dB of dynamic range. This is because of the noise inherent in each system. If you think about it for a second, the quality differences between these systems is obvious. The larger the dynamic range is, the higher the quality of the audio signal and the less apparent any noise is.

Using meters to monitor levels
To control your levels, you need to keep an eye on your meters. Virtually every piece of audio equipment comes with some type of meter to indicate the level of the signal. Meters fall into three main categories: VU meters, LED Peak meters, and software VU/Peak meters, shown in Figure 1.


Figure 1: A software VU/Peak meter Courtesy Sony Sound Forge

The first is VU or Volume Unit meters, which are common on older equipment (and new equipment going for that hip retro look). The needle indicates the overall power of the signal, represented as an average. They're very good for comparing the volume or power of a signal, but not good at registering quick peaks. VU meters usually have two scales, one that runs from 0 percent to 100 percent, and another that has zero where the 100 percent mark is, with negative numbers below 100 percent and positive numbers above 100 percent.

The next type of meter is the LED (Light Emitting Diode) Peak meter. LED meters are very fast, so they are generally used to indicate the peak values of the audio signal. LED meters generally have a single scale, measured in dB, running from approximately -40dB, up through zero, and on to +10 or +20dB.

Note Decibels are a relative measure of power. The decibels used to measure the +20dB measurement on a meter aren't the same as the 120dB pain threshold. One is a measure of sound pressure, while the other is a measure of voltage. It can get kind of confusing

Finally, we have the software meter. Software meters can operate as VU or LED meters, and sometimes as both concurrently. In the image on the right of Figure 1, the meter indicates both VU level (the bulk of the display) and peak level (indicated by the thin line hovering above the VU level). The critical difference between analog meters and software meters is that analog meters have headroom, which means that the signal is allowed to go above zero, and digital meters have no headroom: Go above 0dB, and you'll hear awful square-wave distortion, which is very unpleasant. Digital meters usually include some sort of clip or peak light that indicates when you have peaks above zero. These must be avoided at all costs.

The point of all this is that you need to be careful when setting up your equipment Set your input level too low, and you're liable to hear some of the internal noise of your equipment. Set your level too high, and you'll get distortion. What you want to do is set a level that is high enough so that you don't hear equipment noise and conservative enough that you don't ever get distortion. It's a fairly broad range, so spend the time to learn how to set levels correctly. The result will be a much higher quality podcast.

Setting levels
When setting a level with a VU meter, be careful because VU meters don't register peaks. Those peaks may be loud enough to exceed the equipment's dynamic range and therefore cause distortion. It's best to set your levels between -10dB and -6dB on a VU meter. This leaves quite a bit of headroom for transient peaks, and any unexpected jumps in level. When you're setting levels with a peak meter, you can be a bit more aggressive, because you can see pretty much exactly where your peaks are. You should set your level so that the indicated level is in the -6dB to -3dB range. These settings should get you a good, clean, loud signal, with very little perceivable noise. If your levels occasionally peak above zero, don't worry; most audio equipment has sufficient headroom to handle momentary peaks without distortion.

Setting levels in the digital domain is a whole different matter. Any signal above 0dB causes distortion, because 0dB is considered an absolute maximum. As long as the signal remains above 0dB, you keep getting the same maximum value. The result is a sound wave with the top squared off, which sounds horrible. This is known as square-wave distortion.

You must be conservative when setting levels on digital equipment. Digital meters are almost always peak meters for precisely this reason. When setting your digital levels, you should target -10dB to -6dB. This should leave you plenty of headroom. It's better to be a little conservative and maximize your level later on using signal processing rather than set it too high and end up with distortion.