RealSystem can stream many types of audio in addition to the RealAudio format. Not all audio files stream equally well, however. This chapter describes the types of audio files you can stream, explaining how to prepare or encode your files. It also provides tips for capturing high quality audio source.
The following are tips for creating high quality audio source files. Although geared for RealAudio, these guidelines will help you no matter which audio format you stream.
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Additional Information |
For pointers on video, see "Producing High Quality Video". |
Every piece of equipment in the audio chain, from the microphone, to the sound card, to the sound editing software, affects sound quality. If you intend to provide commercial audio content, invest in professional audio equipment and software. Poor quality equipment can add hiss and distortion, degrading sound clarity.
If you are not recording your own sound, be sure to use high-quality audio source files. Use sources from CD or DAT, for example.
If you are not broadcasting live, capture or "digitize" the sound to a supported file format such as a WAV, QuickTime, or AIFF whenever possible. Digitizing the sound before encoding the file allows you to use a sound editor to adjust the signal amplitude to maximize the available dynamic range. If you do not adjust the signal, the resulting streamed files may sound flat.
Keep the audio source file as small as possible. This makes it easier to encode the file in a streaming format. Cut any unnecessarily long silences from the beginning or end of the file to conserve space.
Setting correct input levels is crucial. All audio equipment has a signal-to-noise ratio, a ratio between the loudest possible sound the equipment can reproduce without distortion and its inherent noise. To work with the loudest input sounds possible, set the input level to use the full range of available amplitude without distortion. This distortion is known as "clipping," and is audible as a high frequency crackling noise.
When digitizing with a sound card, do several test runs and adjust levels on the mixer page of your sound card utilities so the input approaches but does not exceed the maximum. Most mixer pages graphically display input. Make sure there are no peaks above maximum. Be conservative, though. You never know when someone will get excited and speak much louder, or when a crowd at a sports event will roar.
Sound files that do not use the full amplitude range will produce low quality streaming files. If the amplitude range is too low, use your sound editor's Increase Amplitude or Increase Volume command to adjust the range before encoding the file. Most sound editors have a Normalize function that maximizes levels automatically. However, you get better quality if you set the levels correctly when recording.
Eliminate DC offset either while recording content or later with an sound editor. This removes low frequency noise.
Equalization (EQ) changes the tone of the incoming signal by "boosting" (turning up) or "cutting" (turning down) certain frequencies. Using EQ, you can emphasize frequencies you want and cut frequencies that contain noise or unwanted sound. In addition, 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).
As the last step before encoding the file, normalize the source file to 95% of the maximum sound volume. This lets you feed your encoding tool the loudest distortion-free files possible. If your machine's normalization option does not let you specify a percentage, turn down the overall volume after you normalize by using your software's Volume or Amplify option.
If your original audio file signal exceeds the acceptable amplitude range, the file may be "clipped." Clipping can give rise to clicks or pops on playback. If your source file contains a clipped signal, your streaming file may have high-frequency background noise or static. Lowering the input volume helps reduce clipping.
When broadcasting live audio, you have less opportunity to manipulate the input signal. Be sure that volume levels are prepared and tested before encoding live input.
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, with near-CD quality sound at ISDN and LAN speeds. RealAudio files use the file extension .ra.
RealAudio is a compressed format suitable for streaming over low to high network speeds. 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 file from this source file through an encoding tool. Your encoding tool should be able to accept some or all of these input formats:
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Additional Information |
See "RealNetworks Encoding Tools" for more on RealAudio encoding tools. Refer to "Streaming Other Audio Formats" for information on streaming digitized audio files without conversion to RealAudio. |
RealAudio uses a "lossy" compression scheme that disregards parts of the audio source file to achieve a highly reduced file size. A RealAudio file encoded from a WAV file, for example, is typically smaller than the WAV by a factor of ten or more. Although discarding audio information during encoding lowers the file's frequency response and dynamic range, carefully choosing Codecs minimizes the impact of compression.
An encoding tool uses a Codec to compress the original sound file and create a RealAudio file. RealPlayer uses the same Codec to decompress the streamed RealAudio file for playback. When you encode a RealAudio file, you choose a Codec (or series of Codecs) based on two criteria:
As "Targeting Bandwidth" 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.
RealNetworks provides different Codecs for music and spoken voice. Voice Codecs, for example, focus on the standard frequency range of the human voice. Music Codecs have broader frequency ranges to capture more of the high and low frequencies.
When you encode your audio file with a RealNetworks encoding tool, the tool selects the Codec or Codecs automatically. You simply set your bandwidth targets and some other parameters, such as whether you want earlier versions of RealPlayer to be able to play your presentation. The tool then selects the best Codec or Codecs to use based on your input.
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Additional Information |
Refer to the manual or online help for your RealNetworks encoding tool for step-by-step instructions on how to encode a RealAudio file. |
The following table lists all RealAudio Codecs. It is provided as a general reference. Check your RealAudio encoding tool for the Codecs it supports. See below for information about interpreting the table.
The preceding table includes the following information:
An "X" in these columns indicates that a file encoded with this Codec can be played by RealPlayer 6.0, 5.0, 4.0, and so on.
You can use any sampling rate in your audio source file. However, the suggested sampling rates in the table above ensure that the audio stays synchronized with other media and prevents pitch shifting in audio resampling. The highest sampling rate provides the fullest sound.
The following sections give tips for encoding RealAudio files.
You can create a single file encoded for multiple bandwidths only with the Codecs introduced in RealSystem G2. In the preceding table, these Codecs are marked as playable only by RealPlayer 6.0. To support multiple bandwidths when encoding with other Codecs, you must encode a separate file for each Codec. You then use a SMIL file to specify bandwidth choices.
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Additional Information |
For more on bandwidth selection through SMIL, see "Setting Bandwidth Choices". |
When you use a RealNetworks encoding tool to encode a file for multiple bandwidths, you can specify backwards compatibility with earlier versions of RealPlayer. The tool encodes the file for your selected bandwidths with the G2 Codecs. It also includes in the file an encoding at your lowest bandwidth selection that uses an older Codec.
For example, you can encode a single file at 8, 16, and 32 Kbps using the G2 Codecs. In the encoding tool, you can choose backwards compatibility to create an additional 8 Kbps stream with an older Codec. Depending on its connection speed, RealPlayer 6.0 receives the 8, 16, or 32 Kbps G2 stream. Earlier versions of RealPlayer receive the 8 Kbps stream encoded with the older Codec regardless of their connection speeds.
RealSystem can stream audio formats other than RealAudio. Because these formats may not be as highly compressed as RealAudio, they may not be good for low bandwidth connections. The following sections give you guidelines for preparing files in these formats. This information will help you decide if you should stream from the native format or convert the file to RealAudio.
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Note |
The following sections will be added later. |