Chapter 4: Producing Audio

RealNetworks pioneered streaming audio with RealAudio, the first streaming media product for the Internet. Since its debut in 1995, RealAudio has become the standard for network audio, delivering stereo sound over 28.8 Kbps modems, approaching CD-quality sound at higher speeds. RealSystem G2 can stream other audio formats as well. This chapter explains how to prepare and encode your sound files. It also provides tips for capturing high-quality audio.

Steps for Streaming RealAudio

To create a streaming RealAudio clip, you follow these basic steps:

1. Capture audio.

   Unless you start with a digitized audio source file, you capture audio from a voice or music source, such as a person speaking into a microphone or a CD you play through your computer.

   	Additional Information
   	"Recording Audio" provides guidelines for capturing audio.

2. Digitize and optimize audio.

   You next digitize the source audio to a common file format, such as WAV or AIFF. With a sound editor, you can then optimize the audio for streaming. When broadcasting live, however, you encode the streaming audio directly from the source.

   	Additional Information
   	See "Digitizing Audio" for tips on sound editing.

3. Encode RealAudio clip.

   With you digitized file optimized or your live broadcast ready to go, you encode your source in the RealAudio format. When you do this, you choose a codec or set of codecs that target a network bandwidth.

   	Additional Information
   	"Producing RealAudio" explains RealAudio and its codecs.

4. Deliver RealAudio clip.

   With your presentation ready to go, you make your RealAudio clip or broadcast available through your Website. If you are combining sound with another streaming clip, such as RealPix, you write a SMIL file that assembles the pieces.

   	Additional Information
   	Chapter 7 explains how to create a SMIL file. See Chapter 10 for instructions on linking your Web page to a RealAudio or SMIL file. For more on live broadcasting, read Chapter 11.

Recording Audio

Whether you start with existing recordings or capture audio live, the following tips will help you create great audio source files or broadcasts.

	Additional Information
	For pointers on producing video, see "Recording Video".

Use High-Quality Source Media

If you plan to stream existing material, start with the best source possible. Use the cleanest recording with the least amount of unwanted noise. Compact disc (CD) and digital audio tape (DAT) are good source media, although well-recorded analog sources such as records, reel-to-reel tapes, and chrome (type II) cassettes can sound just as good. Try to avoid "consumer grade" recording media such as Type I cassettes and VHS tapes.

Choose Professional Recording Equipment

Every piece of equipment in the audio chain-microphone, mixer, sound card, and so on-affects sound quality. If you intend to provide professional-quality audio content, invest in professional audio equipment and software. Poor-quality equipment can add hiss and distortion, degrading sound clarity.

Use Shielded Cables

It is important to use high-quality, shielded cables. Unshielded cables increase the chance of introducing line noise and Radio Frequency Interference (RFI) into recordings. Keep audio cables physically separated from power cords to minimize the introduction of noise. Also be sure to ground all equipment properly.

Set Input Levels Correctly

Setting correct input levels is crucial. All audio equipment has a signal-to-noise ratio, the ratio between the loudest possible sound the equipment can reproduce without distortion and its inherent noise floor. This distortion is known as "clipping," and is audible as a high-frequency crackling noise.

To get the best signal-to-noise ratio, set the input level on each audio device you use so that it utilizes its full range of available amplitude without distortion during the piece's loudest sections. For example, the audio signal chain may start with the microphone, continue through a small mixing desk, go to a compressor, and finally end at the sound card. For each piece of equipment, set levels as close as possible to 0 dB without going over.

When digitizing with a sound card, attenuate each device to work at its peak efficiency, checking at each point for signal distortion. Perform several test runs and note input levels in your sound editor. Make sure there are no peaks above maximum amplitude. Adjust levels with your sound card mixer so the input approaches but does not exceed the maximum. Be conservative, though. Levels might suddenly increase if, for instance, an interviewee suddenly speaks loudly or a crowd at a sports event roars.

Prepare Volume Levels for Live Broadcasts

When broadcasting live audio, prepare and test volume levels before encoding live input. It is beneficial to have a dynamics compressor (gain compression, not data compression), which is an outboard piece of audio equipment that automatically "rides" the gain level. It maximizes input levels and reduces the chance of distortion by decreasing signal gain that rises above a specified threshold.

Use Optimum RealAudio Sampling Rates

Try to capture sound with sampling width of 16 bits. Note, however, that RealAudio codecs have optimum sampling rates that produce the best sound. The tables in "Choosing RealAudio Codecs" list the recommended sampling rates for each codec.

	Tip
	You do not need to capture stereo sound if you plan to use a mono codec.

Digitizing Audio

If you are not broadcasting audio live, capture and "digitize" the audio to a supported file format such as WAV, QuickTime, or AIFF. Digitizing the audio before encoding a streaming clip lets you edit the audio and maximize its available dynamic range. The following sections give some basic sound editing and optimization tips for digitized audio.

Eliminate DC Offset

DC Offset is extremely low frequency (inaudible) noise that results from equipment grounding problems. If you don't remove it, it can skew the results of subsequent sound editing. Use your sound editor's DC Offset function immediately after recording a digital audio file.

	Tip
	If your sound editing program allows it, eliminate DC offset during recording. This saves you an editing step.

Normalize Audio Files

Set sensible input levels when recording, then use normalization to maximize levels after recording. Your streaming files sound best when your digitized source has the highest possible gain without clipping. Digital audio files that do not utilize their full amplitude range produce low-quality streaming clips. If the amplitude range is too low, use your sound editor to adjust the range and increase the amplitude.

	Tip
	Most sound editors have a Normalize function that maximizes levels automatically.

Equalize Frequencies

Equalization (EQ) changes the tone of the incoming signal by "boosting" (turning up) or "cutting" (turning down) certain frequencies. Using EQ, you can emphasize certain frequencies and cut frequencies that contain noise or unwanted sound. EQ can compensate for RealAudio codecs that do not have flat frequency responses (that is, codecs for which certain frequencies are not as loud after encoding). You can therefore use EQ to make a RealAudio clip sound as close as possible to the initial recording.

Producing RealAudio

RealAudio is a compressed format suitable for streaming over the Internet or intranets. A RealAudio clip can use either .ra or .rm as its file extension. Because RealAudio is compressed, you typically start with a sound file in a digitized, uncompressed format such as WAV or AIFF. You then create a RealAudio clip from this source file through an encoding tool. Your encoding tool should be able to accept some or all of these input formats:

• Audio Interchange Format (.aif)

• Audio (.au)

• QuickTime (.mov)

• Sound (.snd)

• WAV (.wav)

Choosing RealAudio Codecs

RealAudio uses a "lossy" compression scheme that discards parts of the audio source file to achieve a highly reduced file size. A RealAudio clip encoded from a WAV source file, for example, may be 10 to 20 times smaller than the WAV file. Although discarding audio information during encoding lowers the clip's frequency response and dynamic range, carefully choosing codecs minimizes the impact of compression.

A RealAudio encoding tool uses a codec to compress the original sound file and create a RealAudio clip. RealPlayer uses the same codec to decompress the streamed RealAudio clip for playback. When you encode a RealAudio clip, you choose a codec (or series of codecs) based on two criteria:

1. Bandwidth

   As Chapter 3 explains, you need to decide how much bandwidth each part of your presentation will consume. When you have a bandwidth target for your audio component, you can choose a codec that encodes RealAudio at or below that target.

2. Audio Content

   RealAudio uses different codecs for music and spoken voice. Voice codecs focus on the standard frequency range of the human voice. Music codecs have broader frequency response to capture more of the high and low frequencies.

The following tables provide a reference for all RealAudio codecs. Note that your encoding tool may not include all codecs listed. The tables give the following information:

• G2, 5, 4, 3, 2, 1

  An "X" in these columns indicates that a clip encoded with this codec can be played by RealPlayer G2, 5.0, 4.0, and so on. You can encode a single clip for up to four bandwidths with the SureStream codecs, which are playable only by RealPlayer G2. In addition, only RealServer G2 should stream these multiply encoded clips. These clips require a large download time if played back from a Web server.

• Native Rate

  Using a codec's native sampling rate in your audio source file ensures that the audio stays synchronized with other media and prevents pitch shifting in audio resampling. Audio quality degrades if you use lower than the native sampling rate. If you use a higher sampling rate, it is best to use a multiple of the native rate. If the native rate is 8 kHz, for example, use a higher rate of 16 kHz or 32 kHz.

• Resp.

  This column lists the codec's frequency response in kHz.

Encoding RealAudio with RealSystem Tools

When you encode RealAudio clips with a RealSystem G2 encoding tool, you simply set parameters such as audio type (voice or music) and compatibility with earlier versions of RealPlayer. You can also specify multiple bandwidth targets for the clip, such as both 28.8 Kbps modems and ISDN connections. The tool then chooses the best codec or codecs to use. The following sections give tips on using RealSystem tools.

	Additional Information
	See the tool's manual or online help for step-by-step instructions on encoding RealAudio. RealSystem encoding tools are available for purchase or free download at http://www.real.com/products/tools/.

Retain Source Files

Always keep a copy of the original audio source file. To edit the RealAudio clip or encode it with a different codec, modify the source file as necessary, then encode the file again as RealAudio. You cannot convert RealAudio clips to other audio formats.

Using RealAudio in a MultiClip Presentation

When you encode a RealAudio clip, consider whether it will play in parallel with another clip such as a RealPix. If you target 28.8 Kbps modems when encoding, for example, the tool may select a 20 Kbps codec, leaving no bandwidth for the second clip. Make sure you specify that the RealAudio clip is just one part of the presentation. The tool then lets you choose a lower bandwidth codec, such as 8 or 12 Kbps.

Multiple Encoding in a Single Clip

You can create a single RealAudio clip encoded for up to four bandwidths with the SureStream codecs introduced in RealSystem G2. You can also specify backwards compatibility with earlier versions of RealPlayer. The encoding tool then encodes the clip for your selected bandwidths with the SureStream codecs. It also includes in the clip an encoding that uses an older codec and targets the lowest bandwidth choice.

For example, you can encode a single clip at 8, 16, and 32 Kbps using the SureStream codecs. In the RealSystem encoding tool, you choose backwards compatibility to create an additional 8 Kbps stream with an older codec. Depending on its connection speed, RealPlayer G2 receives the SureStream 8, 16, or 32 Kbps stream. Earlier versions of RealPlayer receive the 8 Kbps stream encoded with the older codec regardless of their connection speeds.

	Note
	To support multiple bandwidths with non-SureStream codecs, you must encode a separate clip with each codec. You then use SMIL to specify bandwidth choices. For more on bandwidth selection through SMIL, see "Setting Bandwidth Choices".

Streaming Other Audio Formats

RealSystem can stream several audio formats in addition to RealAudio. The following table lists the streamable formats and shows whether RealPlayer G2 for a certain operating system (Windows 95 or NT, Macintosh, and UNIX) can play the audio. RealSystem typically does not stream these audio formats when the files have been compressed with a codec. Where codec compression is supported, codecs are not included with RealPlayer G2 and must reside on the playback machine already.

	Note
	RealSystem plug-ins may exist for additional audio formats. Check http://www.real.com for information about other audio formats you can stream.

Tips for Streaming other Audio Formats

Observe the following points when streaming audio formats other than RealAudio:

• The audio formats listed above may not work for low bandwidth connections. Test bandwidth consumption by opening the file in RealPlayer G2 and watching the status panel, which will list the streaming bit rate. If the rate is too high, convert the file to RealAudio.

• A-law compression is not supported for any audio format.

• Most uncompressed WAV files work on Macintosh and Unix. Some may not, however. Before streaming a WAV file, test playback locally by opening the file with RealPlayer G2 on these platforms.

Copyright © 1998 RealNetworks
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This file last updated on 09/04/98 at 17:20:32.