Several years ago, one of the most successful radio stations in North America was
CKLW in Windsor/Detroit. Innovative, technically superb audio complemented “The
Big 8”’s excellent execution of their Top 40 program format to give the station a big
signature sound that I have never forgotten. Its striving for perfection has been an
inspiration and has influenced the goals, designs, and sounds of many Orban Optimod products to this day.
If it weren’t for CKLW’s innovative audio texture and the chances that it took to expose the world to many great artists, these discoveries would never have become
part of the radio and music history that they now are. Just as many of these great
artists have moved on from creating hit music to fusing their old styles with newer
forms such as smooth jazz, Orban is evolving by giving the world new viable broadcast technology that build on its legacy.
I dedicate this technical manual to CKLW and those who made it the reality it once
was, and to Leo, my faithful dog, who was beside me, night after night and day after day in the long process of preparing this document.
As the world moves on embracing innovative ways to deliver audio to an audience,
“ladies and gentlemen, the beat goes on…” – Bill Drake, radio programmer
The Adobe pdf form of this manual contains numerous hyperlinks and bookmarks. A
reference to a numbered step or a page number (except in the Index) is a live hyperlink; click it to go immediately to that reference.
If the bookmarks are not visible, click the “Bookmarks” tab on the left
side of the Acrobat Reader window.
This manual has a table of contents. To search for a specific word or phrase, you can
use the Adobe Acrobat Reader’s text search function.
Overview
Opticodec-PC is the first standards-based MPEG-4 AAC/aacPlus™, AAC/HE AAC, ISMA
compliant and SHOUTcast/Icecast compatible encoding software for high quality
streaming audio. Opticodec-PC offers the most important feature that the basic netcaster is looking for in an encoding product — entertainment-quality sound at economical bitrates.
The software lets streaming providers supply content encoded with the Coding
Technologies
available audio quality at the lowest possible bitrate. Streams encoded with Opticodec-PC can be experienced through RealPlayer
ous Ethernet players, and 3G wireless devices. Streams can automatically list themselves on www.opticodec.net
Opticodec-PC offers a choice of a standards-based RTSP/RTP streaming protocol for
use with streaming servers (such as the free enterprise-class, scaleable Darwin
Streaming Server from Apple) or the HTTP/ICY streaming protocol (for use with
SHOUTcast or Icecast Servers). Both server types are non-proprietary and available
for most computer platforms, and some servers are open-source.
Professional radio broadcasters would never consider going on the air without audio
signal processing. They consider it a vital component of the program content, content being what attracts listeners. This carefully crafted content is what holds listeners and keeps them coming back. Broadcast ratings services have proven this true for
over 30 years. Over that period, Orban’s patented Optimod technology has helped
radio and television broadcasters everywhere shape their sound to grab and hold
their listening audiences.
®
AAC/aacPlus codec, widely acknowledged as offering the highest
®
10, QuickTime 6, Winamp 5.05, vari-
, a directory service for Opticodec-PC streams.
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INTRODUCTION ORBAN MODEL 1010
Professional-grade netcasting requires audio processing similar to FM broadcast (although there are some important differences in the peak limiting because of the
different characteristics of the pre-emphasized FM channel and the perceptually
coded netcasting channel). Your listeners deserve to get the best quality and consistency you can provide. Good audio processing is one important thing that separates
the amateur from the professional.
The Orban Optimod-PC 1100, a
professional PCI sound card designed for streaming
media, provides “genuine radio”™ audio processing for Internet broadcasters. With
three on-board DSP's providing mixing, equalization, AGC, multi-band compression,
and look-ahead limiting, Optimod-PC 1100, especially when combined with aacPlus
encoding technology, delivers a polished and produced stream that has the same
loudness, consistency, and punch as satellite and major-market FM radio. In addition
to audio processing, Optimod-PC does internal and external audio mixing, leaving
the CPU power available for encoding with Opticodec-PC. Together, Optimod-PC
and Opticodec-PC provide a unique and tightly tuned system that offers the best
audio quality streams possible with today’s technology.
Opticodec-PC is available in two versions, LE, and PE. Opticodec-PC LE, Light Edition,
is compatible with all quality sound cards and encodes a single stream at bitrates
between 8 and 32 kbps. Opticodec-PC PE, Professional Edition, is offered solely in a
premium package coupled with an Optimod-PC and can encode multiple simultaneous streams at bitrates from 8 to 320 kbps; all streams carry the same Optimod-PC
processed audio content. While the companion Optimod-PC will ordinary be used to
process the stream for consistency and punch, it also comes with presets that allow it
to do simple protection limiting.
Streaming Infrastructure Block Diagrams
AUDIO
INPUTS
Microphone
Pre-A mp
Mixer
Player
Application
PLAYER / ENCODER COMPUTER
ANALOG 1
DIGITAL 1
DIGITAL 2
WAV OUT
WAV IN
OPTIMOD-PC
PCI Sound Card
WAV OUT
Metadata
WAV IN
OPTICODEC-PC
To ServerFrom Encoder
Streaming Audio
Encoder
TCP/UDP/IP
Figure 1-1: Typical streaming infrastructure where program material is sourced from a play-
out system application with live assist
Internet
Network
To Internet
TCP/UDP/IP
SERVER COMPUTER
QTSS/DSS
Streaming Server
OPTICODEC-PC INTRODUCTION
1-3
From
Broadcast
Program Line
AUDIO
INPUTS
ANALOG 1
DIGITAL 1
DIGITAL 2
Metadata
To ServerFrom Encoder
Internet
Network
To Internet
OPTIMOD-PC
PCI Sound Card
WAV OUT
ENCODER COMPUTER
WAV IN
OPTICODEC-PC
Streaming Audio
Encoder
TCP/UDP/IP
TCP/UDP/IP
SERVER COMPUTER
Figure 1-2: Typical streaming infrastructure where program material is sourced from a
radio station on-air studio
OID
UA
STUPNI
ANALOG 1
DIGITAL 1
DIGITAL 2
OPTICODEC-PC
Streaming Audio
Encoder
tenretnI
tenrehtE
redocnE lanretxE morFrevreS lanretxE oT
QTSS/DSS
Streaming Server
tenretnI oT
QTSS/DSS
Streaming Server
WAV INWAV IN
Playout System
WAV OUT
WAV IN
OPTIMOD-PC
PCI Sound Card
WAV OUT
WAV IN
Figure 1-3: Typical multiple streaming encoder/server infrastructure where program ma-
terial is sourced from a player application
Streaming Audio
Streaming Audio
Encoder
Encoder
TCP/UDP/IPTCP/UDP/IP
TCP/UDP/IP
OPTICODEC-PC
OPTICODEC-PC
0008:1.0.0.721 tsohlacol
SHOUTcast
Streaming Server
0108:1.0.0.721 tsohlacol
TCP/UDP/IPTCP/UDP/IPTCP/UDP/IP
Icecast2
Streaming Server
RETUPMOC REVRES / REDOCNE / REYALP
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INTRODUCTION ORBAN MODEL 1010
Opticodec-PC offers the best available tradeoff between audio quality and bitrate.
Compared to MP3, Opticodec-PC provides a better than 60% improvement in audio
quality versus bitrate, reducing network streaming bandwidth requirements and
costs accordingly. At 32 kbps, Opticodec-PC streams offer close to FM quality, without the phasey, watery character of other codecs operating at this bitrate. Many listeners prefer the audio quality of 48 kbps streams to FM.
There is a vast Internet and 3G wireless audience waiting for the entertainmentquality audio that Orban Opticodec-PC and Optimod-PC can provide.
Specifications and System Requirements
PE Version
COMPUTER
Minimum System Requirements:
Windows 2000: Intel® Pentium II 400MHz, RAM = 64MB; 128MB recommended.
Windows XP: Intel® Pentium II 400MHz, RAM = 128MB; 256MB recommended.
Windows 2003 Server Intel® Pentium III 500 MHz
RAM = 256MB; 512 MB recommended
This specification denotes the minimum CPU power necessary to control one
OPTIMOD-PC card with external audio sources and one instance of the OPTICODECPC encoder. Additional cards, audio player and/or encoder software will require additional CPU power.
Processor and Chipset: This software has been tested and qualified with Intel CPUs and
chipsets.
Sound Device: An Optimod-PC 1100 audio processor / sound card must be installed in the
host computer in order to run the Opticodec-PC PE application.
Interface: Graphical User and Command-Line, batchable
Transmission: Automatic Unicast – Announce – Session Description
Protocol (.sdp) file per stream generated and transferred to server
Multicast RTP/UDP (Internal RTSP Server)
TTL: 255 default
Unicast HTTP/TCP
Packet Size: 1450 bytes plus IP Header Bytes = Total < 1500 byte MTU
Connection Fallback: Automatic Reconnection upon Connection Failure
Stream Information: Stream Name and Description; all server supported metadata
Metadata Input: Text File, Serial, Ethernet, Nullsoft Winamp
Server Requirements: Darwin Streaming Server 5.0 and later, QuickTime Streaming
Server 5.0 and later, Nullsoft SHOUTcast DNAS 1.9.4 and later, Icecast2 2.0.2 and later
Server Platform: Available for Microsoft Windows 2000 Professional/Server, Windows 2003
Server, Windows XP Professional, Apple Mac OS X 10.2.8 and later Server and Proxy,
Red Hat Linux 9, FreeBSD, Sun Solaris 9
LE Version
COMPUTER
Minimum System Requirements:
Windows 2000: Intel® Pentium II 400MHz, RAM = 64MB; 128MB recommended.
Windows XP: Intel® Pentium II 400MHz, RAM = 128MB; 256MB recommended.
Windows 2003 Server Intel® Pentium III 500 MHz
RAM = 256MB; 512 MB recommended
This specification denotes the minimum CPU power necessary to control one
OPTIMOD-PC card with external audio sources and one instance of the OPTICODECPC encoder. Additional cards, audio player and/or encoder software will require additional CPU power.
Processor and Chipset: This software has been tested and qualified with Intel CPUs and
chipsets.
Sound Device: Opticodec-PC LE will operate with any Windows-qualified sound card capa-
ble of the required sample rate and bit depth.
Interface: Graphical User and Command-Line, batchable
Transmission: Automatic Unicast – Announce – Session Description
Protocol (.sdp) file per stream generated and transferred to server
Multicast RTP/UDP (Internal RTSP Server)
TTL: 255 default
Unicast HTTP/TCP
Packet Size: 1450 bytes plus IP Header Bytes = Total < 1500 byte MTU
Connection Fallback: Automatic Reconnection upon Connection Failure
Stream Information: Stream Name and Description, all server supported metadata
Metadata Input: Text File, Serial, Ethernet, Nullsoft Winamp
Server Requirements: Free Darwin Streaming Server 5.0 and later, QuickTime Streaming
Server 5.0 and later, Nullsoft SHOUTcast DNAS 1.9.4 and later, Icecast2 2.0.2 and later
Server Platform: Available for Microsoft Windows 2000 Professional/Server, Windows 2003
Server, Windows XP Professional, Apple Mac OS X 10.2.8 and later Server and Proxy,
Red Hat Linux 9, FreeBSD, Sun Solaris 9
These specifications are subject to design improvements and changes
without notice.
Opticodec-PC TE, Test Edition is available upon request for testing encoder/server connectivity. With limited functionality, it allows testing
network connectivity and authentication to verify server configuration.
Applications
Putting your audio content on the Internet or your LAN can be divided into three
main steps: preprocessing the audio signal, encoding it, and streaming it to the network.
High quality streams begin with the cleanest possible audio source material. For best
results, all material should be sourced in digital form to prevent any potential distortion from occurring in the analog-to-digital conversion process. CDs should be digitally extracted (ripped) to a PCM audio format if the digital storage system allows
this, or to a 384 kbps or higher MPEG-1 Layer 2 format. Avoid Layer 3, as well as
other codecs. More information on this topic can be obtained from the Orban publication, “Maintaining Audio Quality in the Broadcast Facility,” available as a free
download from http://www.orban.com.
Preprocessing
For optimum sound, loudness, and peak control, you should digitally preprocess the
Internet audio signal to condition it prior to encoding. The appropriate preprocessing has much in common with the preprocessing required for DAB, HD Radio™, CD
mastering, or digital satellite.
OPTICODEC-PC INTRODUCTION
Preprocessing is necessary for several reasons. Automatic gain control and equalization achieve a consistent sound, while accurate peak control maximizes loudness.
Preprocessing each program element before it is stored on a playout system is not as
effective as preprocessing the mixed audio on the program line immediately before
it is streamed. The latter technique maximizes the smoothness of transition between
program elements and makes voice from, announcers, or presenters merge smoothly
into the program flow, even if the announcer is talking over music.
Peak clipping sounds terrible in digital systems because these systems do not rely on
pre-emphasis/de-emphasis to reduce audible distortion. Instead of peak clipping, the
best sounding processors use some form of look-ahead limiting. The carefully peak
limited signal is then digitally connected to Opticodec-PC to preserve the audio signal waveform integrity.
Orban Optimod-PC (recommended for Opticodec PC LE and required for OpticodecPC PE to operate) is a PCI sound card with on-board digital signal processing that is
suitable for both live streaming and on-demand programming. Its three on-board
Motorola DSP56362 DSP chips provide a loud, consistent sound to the consumer by
performing automatic gain control, equalization, multiband gain control, and peaklevel control. Optimod-PC’s sound card emulation allows it to talk through the operating system via the Windows’ WAVE mechanism to Opticodec-PC, running on the
same computer that houses Optimod-PC.
While there are several types of audio processors available other than Optimod-PC,
conventional AM, FM, or TV audio processors that employ pre-emphasis/de-emphasis
and/or clipping peak limiters are most inappropriate for use with perceptual audio
coders such as Opticodec-PC. The pre-emphasis/de-emphasis limiting in these devices
unnecessarily limits high frequency headroom. Further, their clipping limiters create
high frequency components— distortion—that the perceptual audio coders would
otherwise not encode. None of these devices has the full set of audio and control
features found in Optimod-PC.
Without Optimod-PC processing, audio can sound dull, thin, or inconsistent in any
combination. Optimod-PC’s multiband processing automatically levels and reequalizes its input to the “major-market” standards expected by the mass audience.
Broadcasters have known for decades that this polished, produced sound attracts
and holds listeners.
You can expect a very large increase in loudness from Optimod-PC processing by
comparison to unprocessed audio (except for audio from recently mastered CDs,
which are often overprocessed in mastering). Broadcasters generally believe that
loudness relative to other stations attracts an audience that perceives the station as
being more powerful than its competition. We expect that the same subliminal psychology will hold in netcasting too.
Remote Access & Control:
Optimod-PC has the unique ability to be remotely accessed and controlled over any
TCP/IP network. After the appropriate security and administration setup, OptimodPCs I/O mixer, processing parameters, and presets can be controlled from anywhere,
including from other applications.
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INTRODUCTION ORBAN MODEL 1010
Mixing Facilities:
In addition to sound card and audio processing functionality, Optimod-PC is also a
capable mixer, having one stereo analog input, two AES3 / SPDIF digital inputs
(which can accept any sample rate from 32 to 96 kHz), and one WAVE input (to accept Windows sound sources), all of which can be mixed. Thanks to onboard sample
rate converters, the two digital inputs can accept and mix asynchronous sources,
which may have different sample rates. In practice, the four inputs might be used
for a local feed, a network feed, a voice channel, and a wave player, making Optimod-PC the heart of a “desktop netcasting studio.” In many cases, this versatility
allows you to avoid use of an external mixing desk, thereby keeping the audio path
100% digital. The wave player could be any one of a number of broadcast-oriented
automated playout systems.
Using Optimod-PC’s separate “processed” and “unprocessed” mixers, any of the inputs in any combination can be processed or passed directly to the input of Opticodec-PC without processing—you can always choose how much processing (if any) to
apply to the audio. These features allow local program insertion, such as those required in order to address the broadcast rights issues of many commercials, programs, and events.
Because it uses Microsoft DirectSound Drivers, Optimod-PC is able to play multiple
audio streams from multiple audio sources, eliminating the need for multiple or
multi-channel sound devices for professional playout systems used in automation
mode. Given the CPUs available today, MPEG1 Layer 2, and/or Layer 3 decoding can
occur at the operating system level, eliminating the requirement for expensive
hardware-based MPEG decoder sound devices.
Encoding
Opticodec-PC receives the output of Optimod-PC, which looks like a sound card to
the operating system. Opticodec-PC then reduces the bitrate of the processed signal
by applying it to an AAC or aacPlus perceptual coder and packetizing the resulting
data for an Ethernet network. When the encoder connects to the streaming server,
the encoder generates the Session Description Protocol file and transfers it automatically to the streaming server.
The most basic use of Opticodec-PC is to create a single stream at a single bitrate.
However, the output of a given Optimod-PC card can feed several Opticodec-PC encoders running at different bitrates to service different audience bandwidths; all of
these streams will carry the same audio program.
If you need more than one audio program stream, use multiple Optimod-PC cards
(some of which can be housed in one or more PCI expansion chassis). If you need
multiple streams at different bitrates, configure each Optimod-PC card to feed its
own array of Opticodec-PC PE encoders.
Each installation of Opticodec-PC PE is keyed to one Optimod-PC card, so
running more than one audio program stream requires one Opticodec-PC
PE installation per audio program stream even if all of these installations
are on one computer. However, a single Opticodec-PC installation can
create multiple streams at different bitrates if all of these streams contain
the same audio program.
OPTICODEC-PC INTRODUCTION
About Perceptual Coders
CD-quality audio (16-bit words at 44.1 kHz sample rate) requires 705,600 bits per
second per channel, which is far too high for economical streaming. Perceptual coding reduces the number of bits per second necessary to transmit a high-quality audio
signal.
Perceptual coders exploit models of how humans perceive sound. In particular, perceptual coders exploit the phenomenon of psychoacoustic masking. This means that
louder sounds will “drown out” (or “mask”) weaker sounds occurring at the same
time, particularly if the frequency of the louder sound is close to the weaker sound’s
frequency. Loud sounds not only mask weak sounds occurring simultaneously in
time (spectral masking), but can also drown out weak sounds occurring a few milliseconds before the loud sound starts or a few milliseconds after it stops (temporal
masking).
The basic principle of perceptual coding is to divide the audio into frequency bands
and then to code each frequency band with the minimum number of bits that will
yield no audible change in that band. Reducing the number of bits used to encode a
given frequency band raises the quantization noise floor in that band. If the noise
floor is raised too far, it can become audible and cause artifacts.
A second major source of artifacts in codecs is pre- and post-echo caused by ringing
of the narrow bandpass filters used to divide the signal into frequency bands. This
ringing worsens as the number of bands increases, so some codecs may adaptively
switch the number of bands in use, depending on whether the sound has significant
transient content. This ringing manifests itself as a smearing of sharp transient
sounds in music, such as those produced by claves and wood blocks.
Psychoacoustic Models
Perceptual coders exploit complex models of the human auditory system to estimate
whether a given amount of added noise can be heard. They then adjust the number
of bits used to code each frequency band such that the added noise is undetectable
by the ear if the total “bit budget” is sufficiently high. Because the psychoacoustic
model in a perceptual coder is an approximation that never exactly matches the behavior of the ear, it is desirable to leave some safety factor when choosing the number of bits to use for each frequency band. This safety factor is often called the
“mask-to-noise ratio,” measured in dB. For example, a mask-to-noise ratio of 12 dB
in a given band would mean that the quantization noise in that band could be
raised by 12 dB before it would be heard. (That is, there is a safety margin of two
bits in that band’s coding.) For the most efficient coding, the mask-to-noise ratio
should be the same in all bands, ensuring that the sound elements equitably share
the available bits in the transmission channel.
Increasing the number of bits per second in the transmission always improves the
mask-to-noise ratio. It is important to allocate extra bits to the transmission if the
audio will be processed after it has been decoded at the output of the perceptual
coder (for example, by a second “cascaded” perceptual coder, or by a multiband audio processor such as Optimod-PC). Done correctly, this increased bitrate will raise
the mask-to-noise ratio far enough to prevent downstream processing from causing
the noise to become unmasked.
Because it occurs in narrow frequency bands, unmasked noise does not
sound like familiar white noise at all. Instead, it most often sounds like
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INTRODUCTION ORBAN MODEL 1010
distortion, or like warbling, comb filtering, or gurgling—an “underwater” sound.
Coding Efficiency
Different sounds will vary greatly in the efficiency with which a perceptual coding
system can encode them. Therefore, for a constant transmission bitrate, the maskto-noise ratio will constantly change. Pure sounds having an extended harmonic
structure (such as a pitch pipe) are particularly difficult to encode because each harmonic must be encoded, the harmonics occupy many different frequency bands, and
the overall spectrum has many “holes” that are not well-masked, so that added
noise can be easily heard. The output of a multiband audio processor that uses clipping is another sound that is difficult to encode, because the clipper creates added
distortion spectrum that does not mask quantization noise well, yet may cause the
encoder to waste bits when trying to encode the distortion.
Sophisticated encoders use a short “bit reservoir” to save up unused bits so they can
be applied to difficult-to-encode sounds. However, the length of the bit reservoir
will directly affect the coding delay, so dynamic allocation of bits occurs only over
rather short time windows (in the order of tens of milliseconds). Another feature of
sophisticated encoders is “redundancy reduction,” which encodes frequently appearing data with shorter digital words and infrequently appearing data with
longer words.
Encoding Stereo
Usually, there is some correlation between the left and right channels of a stereo
signal. At lower bitrates, one way to achieve higher quality is to exploit this correlation when coding stereo information:
Depending on program content, the encoder dynamically switches between discrete
left/right coding and sum-and-difference coding. The difference signal often requires fewer bits than the sum signal to encode with high audible quality, thereby
saving bits in the overall coding of the stereo signal.
There is no benefit to joint stereo coding when the two channels contain
independent information because there is no correlation between the
channels.
Opticodec-PC Codecs
Opticodec-PC offers two coding algorithms from the several standardized by
ISO/MPEG (Moving Pictures Experts Group): the AAC and aacPlus® v2 algorithms.
AAC is intended for very high quality coding with compression up to 12:1. The AAC
codec is about 30% more efficient than MPEG1 Layer 3 and about twice as efficient
as MPEG1 Layer 2. The AAC codec can achieve “transparency” (that is, listeners cannot audibly distinguish the codec’s output from its input in a statistically significant
way) at a stereo bitrate of 128 kb/sec, while the Layer 2 codec requires about 256
kb/sec for the same quality. The Layer 3 codec cannot achieve transparency at any
bitrate, although its performance at 192 kbps and higher is still very good.
AAC stands for Advanced Audio Coding. Intended to replace Layer 3, AAC was developed by the MPEG group that includes Dolby, Fraunhofer (FhG), AT&T, Sony, and
Nokia—companies that have also been involved in the development of audio codecs
such as MP3 and AC3 (also known as Dolby Digital™).
OPTICODEC-PC INTRODUCTION
1-11
AAC does not stand for Apple Audio Codec, although Apple was one of
the first to implement this technology with the introduction of Apple
iTunes and QuickTime 6.
The Coding Technologies “Spectral Band Replication” (SBR) process can be added to
almost any codec. This system transmits only lower frequencies (for example, below
8 kHz) via the codec. The decoder at the receiver creates higher frequencies from
the lower frequencies by a process similar to that used by “psychoacoustic exciters.”
Table 1-2: AAC Audio Bandwidth vs. Bitrate, Sample rate, and Channel Mode
OPTICODEC-PC INTRODUCTION
A low-bandwidth signal in the compressed bit stream provides “hints” to modulate
these created high frequencies so that they will match the original high frequencies
as closely as possible. Adding SBR to the basic AAC codec creates aacPlus, which offers the best subjective quality currently available at bitrates below 128 kbps. At bitrates below 128 kbps, full subjective transparency cannot be achieved at the current state of the art, yet the sound can still be very satisfying. (In the phraseology of
the ITU 1 to 5 subjective quality scale, this means that audible differences introduced
by the codec are judged by expert listeners to be “detectable, but not annoying.”)
Coding Technologies’ aacPlus v2, the latest in MPEG-4 Audio and previously known
as "Enhanced aacPlus," is aacPlus coupled with the new MPEG Parametric Stereo
technique created by Coding Technologies and Philips. Where SBR enables audio
codecs to deliver the same quality at half the bitrate, Parametric Stereo enhances
the codec efficiency a second time for low-bitrate stereo signals. Both SBR and Parametric Stereo are backward- and forward-compatible methods to enhance the efficiency of any audio codec. As a result, aacPlus v2 delivers streaming and
downloadable 5.1 multichannel audio at 128 Kbps, near CD-quality stereo at 32
Kbps, excellent quality stereo at 24 Kbps, and great quality for mixed content down
to 16 Kbps and below.
MPEG standardized Coding Technologies’ aacPlus as MPEG-4 HE AAC (MPEG ISO/IEC
14496-3:2001/AMD-1: Bandwidth Extension). With the addition of MPEG Parametric
Stereo (MPEG ISO/IEC 14496-3:2001/AMD-2: Parametric coding for high quality audio), aacPlus v2 is the state-of-the-art in low bitrate open standards audio codecs.
The Coding Technologies codecs provide the absolute best possible sound per bit the
current state-of-the-art will allow, without the typical resonant, phasey, watery
character of other codecs.
Trading-Off Audio Bandwidth against Bitrate, Sample rate, and Channel Mode
High audio bandwidth does not guarantee good sound in codecs. In many cases,
especially at low bitrates, it is actually just the opposite. For example, FM radio is a
15 kHz medium, yet there are plenty of codecs claiming to have 20 kHz response
that sound much worse than FM radio.
1-13
The designers of the various codecs usually determine the optimum tradeoff between bitrate, sample rate, and channel mode (stereo or mono) by performing extensive listening tests. To maximize overall audio quality at lower bitrates, it is important to allocate the bits efficiently. This usually means allocating more bits to
those frequency ranges most important to music and speech.
Below a certain sample rate (which depends on the design of the individual codec),
codec designers have determined that limiting audio bandwidth to less than 20 kHz
achieves highest overall quality. For example, AAC requires 192 Kbps or more for 20
kHz+ response (Table 1-2 on page 1-12) and aacPlus requires 64 Kbps or more for 20
kHz+ response (Table 1-1 on page 1-11).
We recommend using aacPlus v2 for stereo streams below 48kbps. Be sure your target players support it; otherwise, the streams will play in mono.
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INTRODUCTION ORBAN MODEL 1010
Cascading Codecs
There are two general applications for codecs in broadcasting — “contribution” and
“transmission.” A contribution-class codec is used in production. Accordingly, it must
have high enough “mask to noise ratio” (that is, the headroom between the actual
codec-induced noise level and the just-audible noise level) to allow its output to be
processed and/or to be cascaded with other codecs without causing the codecinduced noise to become unmasked. A transmission-class codec, on the other hand,
is the final codec used before the listener’s receiver. Its main design goal is maximum
bandwidth efficiency. Some codecs, like Layer 2, have been used for both applications at different bitrates (and Layer 2 continues to be used as the transmission codec in the Eureka-147 DAR system and many DBS satellite systems). However, assuming use of an MPEG codec, modern practice is to use Layer 2 for contribution only
(minimally at 256 kbps, with 384 kbps preferred), reserving transmission for AAC or
aacPlus. Layer 3 has become a consumer format, and even that is being replaced by
the next generation AAC/HE AAC/aacPlus.
The most general operational advice is this:
•Use compression only when necessary. Hard drives have become very in-
expensive, and there is little excuse for excessively compressing a source library. Linear PCM is best.
• A good codec such as AAC/HE AAC/aacPlus requires a good source to produce
excellent results. If you must use compression in production or transmis-sion ahead of the audio preprocessor (like Optimod-PC) and have the luxury
of high bitrates, use Layer 2 at 128 kb/sec/channel or above (256 kb/sec stereo). This will be audibly transparent for as many as ten passes. Avoid
Layer 3 sources; Layer 3 was never rated transparent at any bitrate.
•Do not use a higher sampling frequency than necessary. 32 kHz is ade-
quate for AM, FM. analog television, and low bitrate (~32 kbps) streaming.
However, if you are creating a hard-disk music library and plan to use it for
DAB or high bitrate streaming now or in the future, 44.1 kHz will yield CDquality bandwidth 20 kHz frequency response. Especially with low bitrate codecs, a 32 kHz sample rate is generally optimum and sounds better than
higher sample frequencies because the bit allocation for the codec is concentrated in the most audible region of the audio spectrum. This is a case where
less is truly more.
•Carefully monitor any cascade of codecs by listening tests. There are
an infinite number of combinations possible, and the human ear must be the
final arbiter of quality. Be particularly sensitive to loss of “snap” and transient
definition, loss of stereo imaging, loss of very high frequencies, comb-filtering
or “underwater” sounds, and buildup of distortion.
• Do not use Microsoft Windows Media Player to play MPEG-1 Layer 2
files. There is a confirmed problem with the MPEG-1 Layer 2 decoder filter
used in the current and several past releases of Windows Media Player. This
filter causes a poor signal-to-noise ratio in the form of low-level noise that is
only there when the least significant bits are present. It is audible during quiet
portions of audio and prevents the filter from being usable in professional
applications. Audio signal processing will make this more apparent. We hope
that Microsoft will someday address this issue.
OPTICODEC-PC INTRODUCTION
1-15
Networking
Opticodec-PC supports both unicast and multicast streams. Each method has its own
advantages and your streaming application will determine which one to use.
To connect to the Internet using unicast, a server is required. This receives the output of the encoder and creates the streams to which your listeners connect. Opticodec-PC supplies an output compatible with the free Darwin Streaming Server, which
is available for multiple platforms including Linux®, FreeBSD®, Sun Solaris®, Microsoft Windows®, and QuickTime Streaming Server for Apple Macintosh®. It is also
compatible with the SHOUTcast DNAS and the Icecast2 servers, also freely
downloadable.
Network Bandwidth Considerations
If you have access to large bandwidth Internet connectivity, you could conceivably
run the server software on your encoder computer—just connect the computer to
an Ethernet Internet feed and you are ready to go. However, most netcasters do not
have that option because the studio or program origination is in one place and the
Internet service provider (ISP) is somewhere else. If that’s the case, the best and most
economical way to connect is to establish what’s called a “co-lo,” or co-location,
which requires running your own server software on another computer, locating
that computer at the ISP, and running one stream per program from your encoder
to the server. Typically, this requires a full-time, non-dial-up dedicated connection
from your encoder to your ISP. Bandwidth requirements for this connection depend
upon the bitrate and number of streams being sent to the server.
A high reliability connection is also recommended to prevent encoder-server disconnects, although Opticodec-PC has the ability to automatically reconnect when this
occurs. If reliability is the goal, avoid consumer Internet connections, especially cable
Internet and some DSL. The relatively small upload bandwidth available from consumer Internet services will severely limit the encoder and/or server. The reliability of
these services is generally not good enough for continuous streaming. Furthermore,
running a server on this type of Internet service may break your Internet service
agreement.
Many ISPs provide servers and administration services to run the appropriate streaming server software. Although you are not responsible for the server administration
in this scenario, it comes at a price.
We have just described how to get your program on the network. Here is where the
listeners come in. There are different ways that people can connect to your stream.
• Most Internet streams are implemented via unicasting, which requires a sin-
gle, independent connection to the server for each stream. (See Unicast on
page 1-24.)
• In a multicast, a single stream is shared among the player clients. Although
this technique reduces network congestion, it requires a network that either
has access to the multicast backbone (otherwise called the Mbone) for content
generally distributed over the Internet, or is multicast-enabled for content distributed within a contained private network. Multicast streams are sent directly to a group address, such an IP multicast address, which many client com-
1-16
INTRODUCTION ORBAN MODEL 1010
puters can simultaneously access. The users of a multicast have no control over
the media content. Multicasts are an efficient way to deliver the same material to a group of people over a LAN, as only one copy of the stream is sent
over the network. (See Multicast on page 1-25.)
Because Opticodec-PC contains a multicast server, more than one listener can
connect to the same IP address without increasing network traffic. This is an
excellent way to deliver corporate or academic content to an internal audience or to stream radio stations to the staff at their computer workstations.
Unless your LAN contains a router that is not multicast enabled and that separates the encoder from your listeners, you do not need to use a server to multicast within a LAN. For listeners to connect to your stream via a typical LAN,
they have to connect their decoder applications to the same local IP address as
the one you assigned to the output of Opticodec-PC.
Bandwidth Requirements
Streaming puts demand on your server system in a number of ways, the most important being bandwidth. For example, three different unicast streams for different
purposes will attract different audiences with different network connectivity requirements.
Stream Type
Distance Learning
Small Corporate Meeting
Medium Entertainment Stream
Large Entertainment Stream
Table 1-3: Bandwidth Requirements for Typical Network Streams
Even the smallest academic streams can generate huge numbers that require more
than a single E-1 or T-1 line to serve. Corporate and entertainment streams can
sometimes require multiple E-3 or T-3 lines, or even higher capacity to serve. Large
streams may even require more than one server to handle the necessary network
throughput. However, since Orban Opticodec-PC is bandwidth efficient, you are
able to serve a larger audience at a lower cost of operation with higher audio quality than with inferior older generation codecs.
Not all networks have 100% of their theoretical capacity available for data transfer.
You are practically limited to about 80% of theoretical maximum because of the
way TCP/IP traffic is handled on a network. For example, a 100 Mbps LAN is limited
to about 80 Mbps. In addition to this practical limitation, you may want to allot additional bandwidth for other tasks, such as file transfers and backup procedures. An
additional 10% should suffice. Generally, a good equation for calculating the practical capacity of a network is:
Practical Network Capacity = Theoretical Maximum * 70%
Maximum Simultaneous Streams = Practical Network Capacity / Stream Bitrate
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