About Shockwave Audio
Shockwave Audio is a technology that makes sounds smaller
and plays them faster from disk or over the Internet.
Shockwave Audio can compress the size of sounds by a ratio
of up to 176:1 and is streamable, which means Director doesn't have to
load the entire sound into RAM before it begins playing. Director starts
to play the beginning of the sound while the rest of the sound is still
streaming from its source, whether coming from disk or over the Internet.
When used properly, the Shockwave Audio compression and streaming features
provide fast playback of high-quality audio, even for users with relatively
slow modem connections to the Internet.
Compression quality in Shockwave Audio
Although Shockwave Audio uses advanced compression technology
that alters original sounds as little as possible, the more a sound is
compressed the more it is changed.
Set the amount of compression by choosing a bit rate setting
in any of the Shockwave Audio Xtras. The bit rate is not related to sampling
rates you may have used in other audio programs. Try compressing the same
sound at several different bit rates to see how the sound changes.
Choose the bit rate appropriate for the intended delivery
system (modem, ISDN, CD-ROM, hard disk, and so on), the type of movie,
and the nature of the sound itself. Voice-over sound quality, for example,
may not need to be as high as that of music. Test the sound on several
systems to find the right balance between quality and performance.
The more compressed a sound is, the faster it streams. If
you choose to use a high quality and low degree of compression, a slow
delivery system may not be able to send the data fast enough, resulting
in gaps during playback. Most developers choose 16 Kbps for the best results
over the Internet.
The following table suggests some general guidelines for
setting the bit rate for different delivery systems. It also provides
a rough estimate of perceived quality for different rates of compression.
Note that real transmission times may be slower than the times shown in
this table, depending on network traffic and server load. Delivery
Bit rate
Quality
T1
64 to 128 Kbps
Equal to source material
ISDN or CD-ROM
32 to 56 Kbps
FM stereo to CD
28.8 modem
16 Kbps
FM monaural or good-quality AM
14.4 modem
8 Kbps
Telephone
Note: Any sound compressed at less than 48 Kbps is converted
to monaural.
Codecs:
-----------------------------
µ-Law 2:1
Useful for exchanging audio with applications on platforms (such as many
UNIX workstations) where µ-Law is a standard audio format. µ-Law is used
for digital telephony in North America and Japan. (The first letter of
the codec name is a Greek letter pronounced Mu.)
16-bit Big Endian and 16-bit Little Endian
Useful when audio must be stored using Big Endian or Little Endian (byte
order) encoding, such as when preparing microprocessor-specific audio.
These codecs are useful for hardware and software engineers but are generally
not useful for video editing.
24-bit Integer and 32-bit Integer
Useful when the audio data must be stored using 24-bit or 32-bit Integer
encoding, such as when preparing microprocessor-specific audio. These
codecs are useful for hardware and software engineers but are generally
not useful for video editing.
IMA 4:1
Useful for cross-platform audio for multimedia. IMA 4:1 was developed
by the IMA using ADPCM.
32-bit Floating Point and 64-bit Floating Point
Useful when audio must be stored using 32-bit or 64-bit floating point
encoding, such as when preparing microprocessor-specific audio. These
codecs are useful for hardware and software engineers but are generally
not useful for video editing.
ALaw 2:1
Similar to µ-Law, but used primarily for digital telephony in Europe.
MetaSound/MetaVoice Codecs (Mac OS only)
A wide series of codecs developed by Voxware. These codecs discard parts
of the audio signal that are imperceptible to the human ear so the compression
provides high music quality with high compression. The codecs cover a
wide range of bit rates from AC06 V2.0 at 6,000 bps to the ACS96 V2.0
at 96,000 bps i order to accommodate varying bandwidths.
Qualcomm PureVoice
Intended for speech; works best at 8 kHz. Based on the Code Division Multiple
Access (CDMA) technology standard for cellular telephony.
QDesign Music Codec
Useful when compressing high-quality music for Internet distribution.
It is capable of delivering CD-quality (16-bit, 44.1 kHz) audio over a
28.8 Kbps line.
MACE 3:1 and MACE 6:1
Useful as a general-purpose audio codec. The Macintosh Audio Compression
and Expansion codec (MACE) has been built into the Mac OS Sound Manager
for many years. MACE 3:1's lower compression ratio provides higher quality
than MACE 6:1. Because it is provided with QuickTime 3.0 and later, it
is also accessible in Windows when QuickTime is installed.
MPEG Layer-3 Codec
Also known as MP3. This is the third coding scheme for MPEG audio compression.
MPEG Layer-3 uses perceptual audio coding and psychoacoustic compression
to remove parts of the audio signal that are imperceptible to the human
ear. The result is a compression ratio up to 12:1 without loss of audio
quality. MP3 is a common format for distributing music files over the
Internet.
ACELP.net
A net-based codec using frame-concatenation and interlacing for improved
music quality. ACELP.net allows a dual-rate bit-rate of 8.5/6.5 kbps or
a fixed-rate bit-rate of 5.0 kbps.
WM-AUDIO
More fully known as Microsoft® Windows Media™ audio compression. This
is the standard codec for Microsoft Active Streaming Format which combines
fast encoding with high music quality and is optimized for Pentium II
(MMX) and Pentium III (SSE/SIMD) processors. WM-AUDIO has a wide bit-rate
range from 5 kbps to 128 kbps and offers high quality sound over the Internet
even over 28.8 modems. WM-AUDIO is considered a future replacement for
MP3.
Indeo® Audio Software
Useful for music and speech distributed over the Internet. Its maximum
compression ratio is 8:1. This codec is designed to work together with
the Indeo Video codec.
Microsoft G.723.1
A codec intended for use in video conferencing. It offers acceptable voice
quality, but is a poor choice for music or sound effects. The audio quality
is lower than other codecs that use the same data rate.
L&H Codecs
Speech and music compression algorithm developed by Lernout & Hauspie™.
TrueSpeech™
Useful for speech over the Internet at low data rates.
Microsoft GSM 6.10
Useful for speech, used in Europe for telephony.
Microsoft CCITT G.711
This codec uses µ-Law encoding and is commonly used for digital telephony
in North America and Japan.
MS-ADPCM
A Microsoft implementation of Adaptive Differential Pulse Code Modulation
(ADPCM), a common digital audio format capable of storing CD-quality audio.
Microsoft IMA ADPCM
An implementation of ADPCM, useful for cross-platform audio for multimedia,
developed by the Interactive Multimedia Association (IMA).
3D sound
http://www.hitl.washington.edu/scivw/EVE/I.B.1.3DSoundSynthesis.html
Soud Synthesis info
Sound Forge 4.5
http://www.sonicspot.com/soundforge/soundforge.html
List of Real Time Softare Synthesizers
for PC
http://www.sonicspot.com/softwaresynth.html
Sound synthesis - general
http://www.sfu.ca/sonic-studio/Sound_Synthesis.html
Sound synthesis by physical
modeling - IRCAM
http://viswiz.gmd.de/~eckel/publications/eckel95b.html
Physical Modeling
http://www.audionica.com/Audionicaen/synthes.htm
http://viswiz.gmd.de/~eckel/publications/eckel95b.html
(IRCAM)
http://www.dei.unipd.it/english/csc/documents/pm/pm.html
http://web.ukonline.co.uk/taosynth/
Sound synthesis by physical
modeling - Italy
http://www.dei.unipd.it/english/csc/documents/pm/pm.html
Fourier Synthesis:
http://www.phy.ntnu.edu.tw/java/sound/sound.html
Chaotic/Fractal synthesis
http://www.far-field.com/~dan/Music/chaos/Chaosrel.htm
http://www.ecdl.hut.fi/~kimmo/emusic.html
Chaotic sound synthesis
http://www.far-field.com/~dan/Music/chaos/Chaosrel.htm
http://www.sonicspot.com/guide/synthesis.html
Genetic Algorithms
http://www.cs.ukc.ac.uk/pubs/1999/908/
?
http://datura.cerl.uiuc.edu/BillWalker/thesis/thesis.html
online voice/speech synthesis
A research version of Next-Generation Text-To-Speech (TTS) from AT&T
Labs
http://www.research.att.com/~mjm/cgi-bin/ttsdemo
http://www.helsinki.fi/~ssyreeni/dsound/index.html
Fourier synthesis (additive
synthesis)
http://www.phy.ntnu.edu.tw/java/sound/sound.html
http://www-users.york.ac.uk/~mdjp101/bongsmack/
CSound
http://www.csounds.com/
CSound for MAC
http://music.columbia.edu/~matt/
Composers
Paul Lansky
http://www.essentialsofmusic.com/composer/lansky.html
Xenakis
http://elib.zib.de/ICM98/C/1/urania/abstracts/Hoffmann.html
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