Audio Codecs for Internet Streaming
A & M I n t e r a c t i v e / D i s t r i b u t i o n T e c h n o l o g i e s
A u t u m n 2 0 0 6
D i s c u s s i o n D r a f t
Version C.3 / October 30th – March 22nd 2007
Check for updates here:
http://audioandmusic.gateway.bbc.co.uk/departments/rmi/AOPAPER_AudioCodecsForInternetStreaming/
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Executive Summary
The purpose of this paper is to discuss common formats for streaming audio, informing the reader of important factors
that guide our adoption of new formats and to provide recommendations.
Audio codecs are hardware or software processes which allow us to transform an audio signal into something suitable
for digital transmission over the internet, digital radio, digital TV, etc.
It is important to BBC Audio & Music (A&M) that we stay informed to keep our audience’s access to our content as
wide as possible. Changing business practices in the world of audio codecs have a direct impact on how we should be
delivering our content, and it’s not necessarily the best quality codec that wins.
The listening quality of the codecs is not of primary focus here, but rather the business logic surrounding them – this
includes uptake of codecs by software players, uptake of codecs by manufacturers of embedded audio devices (e.g.
WiFi Radio, Networked Media Appliances), pressure for adoption of non-proprietary codecs and pressure from content
providers to protect their audio content using Digital Rights Management (DRM).
We cover important ‘buzz-words’ surrounding audio codec discussions by way of definitions with examples to aid
understanding.
The BBC’s Streaming Strategy and Audio/Video Standards groups delivered the strategy for our present situation: ‘dual
format’ codec use as a minimum; Windows Media Audio and Real Media Audio. We look at whether this is still valid,
and if any changes are required.
The main body of the paper presents our coverage of a group of common Audio Codecs including key assumptions:
•
Advanced Audio Coding (AAC)
•
AACPlus (High Efficiency AAC, AAC+, HE-AAC)
•
MPEG audio layer-3 (MP3)
•
Real Audio (RA)
•
Vorbis (Ogg Vorbis / Xiph Vorbis, Ogg)
•
Windows Media Audio (WMA)
•
Adaptive Multi-rate (AMR Narrowband) / Adaptive Multi-rate Wideband (AMR-WB) / Adaptive Multi-rate
Wideband Plus (AMR-WB+, AMR-WBPlus)
Our conclusions deliver recommendations for adoption of AACPlus as a third supported codec for our internet
services and how we may go about providing this on new Multicast and Unicast services, by means of a trial. We give
a balanced view of supporting MP3 as a possible streaming format, but do not recommend supporting for anything
other than specialist uses with clear business cases. We also recommended that the BBC should form closer
relationships with the ‘major players’ in this area – Apple, Microsoft, Real Player, Adobe.
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Contents
Introduction .....................................................................................................................4
Definitions ........................................................................................................................5
Codec
5
Container
5
Streaming Media
5
Progressive Download Media
6
Bit rate
6
Sample rate
6
Lossy vs. Lossless vs. Uncompressed
6
Digital Rights Management (DRM)
7
Present situation ..............................................................................................................8
Codecs ............................................................................................................................9
Advanced Audio Coding (AAC) 10
AACPlus (High Efficiency AAC, AAC+, HE-AAC) 10
MPEG audio layer-3 (MP3)
11
Real Audio (RA)
11
Vorbis (Ogg Vorbis, Xiph Vorbis, Ogg) 12
Windows Media Audio (WMA) 12
Adaptive Multi-rate Family (AMR) 13
Summary
13
Conclusions ...................................................................................................................14
Appendix A: Codecs matrix ...........................................................................................15
Appendix B: Codecs not covered..................................................................................19
Appendix C: Assumptions .............................................................................................21
Appendix D: Current Bit-rates .......................................................................................23
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Introduction
The purpose of this paper is to discuss common formats for streaming audio, informing the reader of important factors
that guide our adoption of new formats and empowering the reader with a grasp of the technical terms used.
Audio codecs are hardware or software processes which allow us to transform an audio signal into something suitable
for digital transmission over the internet, digital radio, digital TV, etc.
It is important to BBC Audio & Music (A&M) that we stay informed to keep our audience’s access to our content as
wide as possible. Changing business practices in the world of audio codecs have a direct impact on how we should be
delivering our content, and it’s not necessarily the best quality codec that wins.
This paper is a discussion document, and contains many assumptions. These assumptions have been detailed in the
appendices and will be raised up-front where appropriate. Discussion is important, and the complexity of the issues
surrounding audio codecs you may find you question – it is important that you raise these concerns so that we can
revise the paper on a regular basis.
The topics dealt with in the paper are constantly changing in many areas so what is valid today may be invalid
tomorrow, it is important that Audio & Music stay informed and this paper is designed to do that.
This paper is arranged in the following way:
•
Definitions: In order to give you some definitions of common words used when discussing Audio Codecs, we
begin with a quick round-up of definitions.
•
Present Situation: Then we have an overview of the current A&M, and pan-BBC, strategy for Audio Codec
use.
•
Codecs: Followed by information on our reviews process and our findings, this is backed up by Appendix A
which shows the full detail of our findings and Appendix B which lists Audio Codecs which we didn’t review.
We also cover some key assumptions we made during our process, and a full list of these is available in
Appendix C.
•
Conclusions: Finally, we present our conclusions with recommendations on next steps.
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Definitions
We have collated a few common terms and their definitions which will help understanding this paper. Some of these
terms are misused in general discussions and it is important that we use them correctly in this paper.
Codec
A codec (‘CODer – DECoder’) is a process which is capable of encoding or decoding an input signal. Codecs can be
implemented in software or hardware, and act on a digital or analogue input signal. They encode a signal for
transmission or storage and they decode it for playback or editing. Encoding can use compression on the input signal
and results in smaller file sizes of the resultant output (please see ‘Lossy vs. Lossless vs. Uncompressed’ below for a
further discussion about compression). A codec family is where a group of codecs share the same algorithms for
encoding but can encode to different bit rate and sample rate combinations (see below). Example - an input audio
signal could be encoded to be used for transmission on the internet, an input video signal could be encoded to make
the storage requirements for the content smaller. Some example codecs would be Windows Media Audio, MPEG Audio
Layer 3 (MP3), and MPEG 4 Advanced Audio Coding (AAC).
REF:
http://en.wikipedia.org/wiki/Codec
Container
Often used interchangeably with the codec’s name, the container merely specifies the way that packages of encoded
signals can be handled together, they can consist of 1 or more encoded signals. A common misconception is that if a
device/player is able to handle a specific container, then it can handle any codecs used for the signals contained inside
– this can be incorrect, and results in the common problem where a device/player supports a container but not the
codec required to playback one or more of the signals contained inside. Some examples of containers would be AVI
(Audio Video Interleave), MOV (a standard Quicktime container). This paper discusses codecs rather than the containers
which they could fit in, though all of the codecs we have reviewed fit into the standard containers which are common to
them so we need not concern ourselves with the container types at this level of discussion.
REF:
http://en.wikipedia.org/wiki/Container_format_%28digital%29
Streaming Media
The traditional description of streaming media is audio/video transmitted over the internet that can be viewed before it
is buffered in its entirety. It is used for Live and On-demand content. However there can be ambiguity when compared
with the newer technique of ‘progressive download’ (see below). To distinguish it is necessary to say that Streaming
Media is delivered in such a fashion that the Player can drop frames (or parts of frames) during playback in order to
keep the playback synchronised with the server that is delivering the stream. There are different protocols for
streaming, traditional Web Browsing using HTTP (Hyper Text Transfer Protocol) is unable to support the features
required. HTTP delivers every piece of a requested piece of content, be that an HTML page or an Image, this doesn’t
work with streaming. RTSP (Real Time Streaming Protocol) is a common protocol used for streaming – this was based
on HTTP but gives the flexibility that allows us to control the playback, jump to any point in an on demand stream and
gives the ability to throttle the bandwidth in some formats by providing a multiplexed stream (multiple bit-rate stream)
to give a service which works on variable bandwidths. Streaming servers require special streaming enabled software to
work, a standard Web Server can not deliver Streaming Media. Examples of common Streaming Server applications
are: Windows Media Services, Real Server / Helix Universal Server, Quicktime Broadcaster. A regular consumer of
streaming media would find in non-trivial to capture and save Streaming Media.
REF:
http://en.wikipedia.org/wiki/Streaming_media
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REF:
http://en.wikipedia.org/wiki/Rtsp
Progressive Download Media
This technique is where media is delivered via regular HTTP (see above for comparison) using a regular Web Server.
This means, unlike Streaming Media that play position cannot be moved to an offset in the media that has not been
downloaded. Download of the media using this technique is linear, i.e. the frames of audio/video are downloaded in
order and contiguous. However, most players that work with Progressive Download media allow for playback before
the entirety of the media is downloaded. You commonly see this with services like Google Video which uses Flash
Video delivered using Progressive Download, and can be seen where you cannot move the playhead on the video past
a buffer point that shows how much data for the media has been downloaded in the background. Progressive
Download media is easy to save from your cache, so the regular consumer can easily capture your media. It is often
also more desirable where requirements are to deliver higher-quality video that bandwidth changes can make it difficult
to deliver the same quality using a streaming server.
REF:
http://www.streamingmedia.com/article.asp?id=8456
Bit rate
During analogue to digital encoding or during conversion of one previously encoded file to another codec encoded file
the bit rate represents the amount of data stored about the audio per each time unit. Bit rate is measured in number of
bits per second - bps, in audio codecs this is usually at the thousands of bits (kilo-bits) per second rate: kbps, or
higher. (see first web reference below for other rates). The bit rate is affected by a number of factors:
•
the sampling rate (see below) of the input signal
•
the codec’s internal encoding algorithms
•
compression within a codec’s algorithm
To complicate things a little there are also variable bit-rate and constant bit-rate versions of some codecs. Most
common are constant bit-rate codecs where consumption of the data by a decoder is handled in a constant rate,
variable means that the consumption happens in varying rates. Variable bit-rate usually out-performs the quality of
constant bit-rate output, though adds extra complexity to decoders in players.
REF:
http://en.wikipedia.org/wiki/Bit_rate
REF:
http://en.wikipedia.org/wiki/Constant_bitrate
REF:
http://en.wikipedia.org/wiki/Variable_bit_rate
Sample rate
During conversion from an analogue input source, or from one previously encoded file to another, the sample rate
defines the number of samples taken of the signal per second. It is measured in Hertz (Hz). In simple terms – the higher
the sample rate, the more accurate the preservation of the input signal quality prior to the codec’s encoding and
compression algorithms.
REF:
http://en.wikipedia.org/wiki/Sample_rate
Lossy vs. Lossless vs. Uncompressed
Three terms are commonly used in the description of any codec and they all relate to whether compression is used or
not and how much compression is used on the input signal during processing. Working out how much compression
should be used, or if it should be used at all, is a balance between required file size, quality of audio and complexity to
decompress during playback. Compression ‘artefacts’ due to characteristics of a codec’s compression algorithm affect
the resultant output.
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Uncompressed : the codec uses no algorithms during the encoding process which compresses the input
signal, though there will notionally be some amount of data loss during the analogue to digital process or from
one codec encoded file to another. File sizes are usually very large when using high sample rates. Common
uses: for handling audio for broadcast or input to another encoder. Example codecs: WAV, BWAV.
•
Lossless : the codec uses a compression algorithm that allows ‘exact’ reconstruction of the input signal
during decompression and playing. File sizes are still reasonably large. Common uses: ripping audio CD tracks
to preserve the quality, making a file smaller to distribute it more efficiently (than without compression).
Example codecs: FLAC, Monkey’s Audio. REF:
http://en.wikipedia.org/wiki/Lossless
•
Lossy : the codec uses a compression algorithm which has a trade off between audio quality against file size,
this balance can usually be controlled by the user during set up of the encoder. The quality of the algorithm for
compression is directly linked to the resulting output quality, a lesser algorithm usually results in poorer
quality… a cleverer algorithm tends to have better quality. Common uses: ripping audio CD tracks when file
size is more important over quality, streaming media. Example codecs: MP3, Real Audio, Windows Audio.
REF:
http://en.wikipedia.org/wiki/Lossy_data_compression
Digital Rights Management (DRM)
Publishers and copyright owners of media often use DRM to protect their intellectual property. A number of DRM
software technologies exist and have to be embedded into the players of the content in order for consumers to be able
to unlock the content and listen to it. This often reduces the number of players that support playback of content, as it is
no longer a question of supporting just the correct Audio Codec. Applying DRM can also limit which platform’s can
playback the audio, and on some it limits it completely (e.g. open source based operating systems, e.g. Linux).
Common DRM features include:
•
encryption of the media file itself, or encryption of the container the media file is in (‘envelope DRM’)
•
rules that determine access control to playback the piece of media (e.g. embargo - playing stops after a
particular date/time, preload - playing is only available after a particular date/time, duration - playing is only
allowed for a given duration, listens - playing is only allowed a certain number of times, etc)
•
rules that determine control over usage of the piece of media (e.g. burning to CD, copying to portable media
device)
DRM can be a controversial issue, and you should really check out the link below if you want to learn more.
REF:
http://en.wikipedia.org/wiki/Digital_Rights_Management
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Present situation
The current BBC standards for Audio distribution on the Internet are currently based on two main codec families:
•
Windows Media Audio – developed by Microsoft. Our live audio streams are encoded at Stereo 48kbps,
currently we have no Audio on Demand encoded but it is planned for 2007. On most of our streams we use
Windows Media’s multiple bit-rate streaming technique – Intelligent Streaming.
•
Real Media Audio – developed by Real Networks. Our Live and Audio on Demand streams are encoded at a
maximum of Stereo 44kbps. We use Real’s multiple bit-rate streaming technique – SureStream.
[A full breakdown of the streaming rates of our Live and Audio on Demand streams is available in Appendix D]
This is our ‘dual format’ minimum standard that we adhere to, its aim was to cover the most audience possible whilst
being cost efficient for delivery. By covering Windows, Mac OS X and Linux/Unix platforms to greater or lesser extents
we cover as many of our audience as possible. This is an entirely valid decision to have made, as you will see when we
compare the codecs below.
However, we now have to consider that our content won’t only be required by our audience on their home/office
computers, with the advent of manufacturers delivering mobile phone handsets with multimedia playback capabilities,
Digital Audio Broadcasting (DAB) - digital radio, networked media appliances (e.g. WiFi Radios) and Digital Radio
Mondiale we need to consider if the dual format concept is valid.
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Codecs
The audio codecs we have chosen to review are (common:
•
Advanced Audio Coding (AAC)
•
AACPlus (High Efficiency AAC, AAC+, HE-AAC)
•
MPEG audio layer-3 (MP3)
•
Real Audio (RA)
•
Vorbis (Ogg Vorbis / Xiph Vorbis, Ogg)
•
Windows Media Audio (WMA)
•
Adaptive Multi-rate (AMR Narrowband) / Adaptive Multi-rate Wideband (AMR-WB) / Adaptive Multi-rate
Wideband Plus (AMR-WB+, AMR-WBPlus)
The detail we have collected on these codecs can be found in Appendix A. Since our primary focus is on streaming
audio, we have ignored Uncompressed and Lossless codecs since their properties do not make them desirable for
streaming. We have also ignored some less common, or older, Lossy Codecs since their support in players is low or
have been super-seeded by other codecs. (A list of the codecs we have ignored is available in Appendix B)
The listening quality of the codecs is not of primary focus here, but rather the business logic surrounding them – this
includes uptake of codecs by software players, uptake of codecs by manufacturers of embedded audio devices (e.g.
WiFi Radio, Networked Media Appliances), community pressure for adoption of non-proprietary codecs, pressure from
content providers to protect their audio content using DRM.
Consider the tale of VHS versus BETAMAX. Even though BETAMAX was technically better, VHS won out due to the
business factors (more manufacturers embedded VHS systems because it was cheaper than BETAMAX, though VHS
tapes wore out far quicker than BETAMAX and were inferior technology). This is called ‘to BETAMAX’, where a
technology that was technically better lost out due to market forces. (If you are interested in the full story of BETAMAX
see:
http://en.wikipedia.org/wiki/Betamax)
We have attempted to provide some 'feel' for the codecs listening qualities, which we will summarise below and part of
which is available in the column marked 'Quality vs. Filesize (Bandwidth)' in Appendix A’s spreadsheet – this is purely
subjective but gives some idea of the issues.
It is also vitally important to understand the assumptions we have made during our testing processes – a full list is
available in Appendix C, but some of the key assumptions we have made are:
•
Flash Video (FLV) describes a container which has a Video Stream and an Audio Stream, the only codec
supported for the Audio Stream is MP3. This means that Flash Video does not appear as a codec that we
tested, however Flash Player does appear in our tested players.
•
Windows Media Player 10 Mobile, which is embedded on Windows Mobile based devices (e.g. XDA, Orange
SPV, Pocket PC devices, etc) does not seem to support 3GPP standards for audio, only Windows Media
Audio and Windows Media Video, MP3 and ISO MPEG-4 video. Here we have therefore treated it as
supporting either Windows Media Audio or MP3.
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•
Playstation 2, and similar ‘games consoles’ have games created for them by the games developer, it is up to
that developer to decide which audio codec they wish to include libraries for. In a couple of the examples we
have seen of Playstation 2 games which allow internet streaming media to be played, they have been MP3
streams. The use of different libraries to support different audio codecs is up to the games developer at time of
development. Some of these games consoles now have an interface outside of the games which also supports
some specific codecs, however information on these is somewhat limited by lack of relationships with games
console manufacturers.
Advanced Audio Coding (AAC)
Offering excellent audio quality, this codec has similar filesize to MP3 but significantly better quality to filesize ratio at
the same bit-rates. Real Audio even drops to AAC encoding for any audio encoded over 96kbps, hence it is also
natively supported in Real/Helix.
•
Codec has excellent cross-platform capabilities (PC/Mac/Linux/Mobile and other devices)
•
Multi-bitrate streaming is not possible.
•
DRM is possible with Helix DNA DRM, and iTunes Fairplay. It does not have any native DRM technique.
•
AAC has good coverage of codecs shipped in common software players (see list in Appendix A); only
Windows Media Player requires a plug-in to allow it to playback.
•
AAC works on 10/15 of the embedded devices and other systems covered in our tests (see list in Appendix A).
Potential of codec: Since AAC is part of the 3GPP specifications, we can see widespread potential usage on mobile
handsets. iTunes supports AAC natively and accounts for a high number of users, though not as high in number as
Windows users able to playback WMA.
AACPlus (High Efficiency AAC, AAC+, HE-AAC)
There are currently two versions of the AACPlus codec, and here we are assuming that the latest version is used, at
time of writing this is version 2 – AACPlus v2. Please note that when talking about AACPlus in this document we are
referring to this version.
Excellent audio quality with lower filesize than AAC, it performs much better at the lower bitrates circa-56kbps than
Real Audio or Windows Media Audio. A small increase to 64kbps results in very good audio, 96kbps and it’s excellent.
AACPlus is designed to allow codecs that can decompress AAC to do so, but the loss of quality is noticeable. Only a
native AACPlus decoder can output audio at its full potential.
•
Codec has excellent cross-platform capabilities (PC/Mac/Linux/Mobile and other devices)
•
Multi-bitrate streaming is not possible.
•
DRM is possible with Helix DNA DRM (though not currently demonstrated, it is part of the specification). It
does not have any native DRM technique.
•
AACPlus has reasonable coverage of codecs shipped in common software players (see list in Appendix A),
Windows Media requires a plug-in and it’s likely that next versions of the other players will have the codec
shipped in them.
•
AACPlus works on 7/15 of the embedded devices and other systems covered in tests (see list in Appendix A).
•
Age of codec: New
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Potential of codec: worldDMB has adopted AACPlus as an additional codec for the DAB standard, it is also the codec
adopted by Digital Radio Mondiale. AACPlus is also part of the 3GPP specification, so mobile devices will support it.
Some Internet Radio stations are already delivering an AACPlus stream in addition to other formats or as an alternative.
We feel that there is a movement towards the adoption of AACPlus with in the ‘Jukebox-style’ Internet Radio stations,
as the clear benefits of lower bandwidths and improved quality are seen by the owners as an improvement to their
business model – this should also act as another influencing fact or media player’s roadmaps.
MPEG audio layer-3 (MP3)
Encoder efficiency is worst out of others compared in this document. MP3 needs high bitrates to come close to AAC
quality at lower bitrates. MP3 technically requires a commercial license, though there are some less than efficient ‘open
source’ solutions.
•
Codec has excellent cross-platform capabilities (PC/Mac/Linux/Mobile and other devices)
•
Multi-bitrate streaming is very unlikely. There is some documentation about using a multi-bitrate stream for
MP3; however we cannot find a player that mentions support of this during playback.
•
DRM is possible with Helix DNA DRM. It does not have any native DRM technique. It is notable that MP3 is
one of the easiest audio codecs to record, and ‘save to disk’ was something that the music industry picked up
on a few years ago and is probably their reason for disliking MP3.
•
MP3 has excellent coverage of codecs shipped in common software players (see list in Appendix A), none of
our tested players required a plug-in to playback MP3
•
MP3 works on 10/15 of the embedded devices and other systems covered in our tests (see list in Appendix A).
Potential of codec: This is an old codec, with an algorithm that doesn’t compare with advances like AAC/AACPlus and
some of the improvements in Real Audio at lower bitrates. This format is currently very common for Internet Radio
stations – however you should consider the comments above in the potential of AACPlus codec as it starts to influence
MP3 codec use.
Real Audio (RA)
Very good at low bitrates – low bitrates provide excellent quality for file size ratio. Codec quality is better than Windows
Media. At bitrates higher than 96kbps the codec switches to AAC from its Real-native encoder – as a definitive nod that
AAC would be better than their codecs at that rate.
•
Codec has excellent cross-platform capabilities (PC/Mac/Linux/Mobile and other devices)
•
Multi-bitrate streaming is handled by Real’s proprietary ‘Surestream’ technique which requires streams to be
placed on a Real Media compliant streaming server and played with a Real Media compliant player (e.g. Real
Player or Helix Player). Surestream works well, possibly due to its maturity in the marketplace.
•
DRM is possible with Helix DNA DRM, which is also Real Audio’s native DRM technique. Even without DRM
applied, streams are somewhat difficult for the average consumer to record – requiring a specialist piece of
software (not common) or intimate knowledge of their computer’s Sound Card configuration.
•
Real Audio has poor coverage of codecs shipped in common software players (see list in Appendix A); none of
our other tested players had available plug-ins to playback Real Audio. Older versions of the codec do appear
in a plug-in called the ‘Real Alternative’, however we are using the latest codecs to encode our streams so it
will not play in that plug-in.
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•
Real Audio works on 1/15 of the embedded devices and other systems covered in our tests (see list in
Appendix A).
Potential of codec: with the life of this codec extended by the fact that Real Networks spun off development of an
open source player (Helix Client) that is embedded in many hardware devices, including mobile handsets, and Real’s
open admission that at higher bitrates (>96kbps) they can’t do better than AAC. This does mean that the potential of
the codec is still important but for higher bitrates it’s pointless. The proprietary nature of the libraries in the open source
player is annoying though as the Helix Client works with libraries to provide codecs – unfortunately the libraries for Real
codec aren’t entirely open which is a big problem on Linux.
Vorbis (Ogg Vorbis, Xiph Vorbis, Ogg)
Originally designed as a lossless codec, the developers of Vorbis switched to make this codec lossy at bitrates lower
than 96kbps. Very good at all bitrates - low bitrates are excellent quality for filesize though filesize is larger than
Windows Media or Real Audio. Codec quality is better than Windows Media. Overall sound not as good as AACPlus
though.
•
Codec has reasonably good cross-platform capabilities (PC/Mac/Linux)
•
Multi-bitrate streaming is not possible.
•
DRM is possible with Helix DNA DRM. It does not have any native DRM technique.
•
Vorbis has reasonable coverage of codecs shipped in common software players (see list in Appendix A), plug-
ins were available for most other players we tested.
•
Vorbis works on 5/15 of the embedded devices and other systems covered in our tests (see list in Appendix A).
Potential of codec: Vorbis is still an oddity, even though it seems to be a good codec its support on embedded
devices and some players is still unacceptable despite the fact that it’s an Open Source codec and it can freely be
embedded. It’s difficult to tell why it hasn’t received mass support, but now it’s superseded by AAC/AACPlus so it’s
unlikely we’ll see improved support.
Windows Media Audio (WMA)
Low bit rate is problematic. However it redeems itself at higher bitrates. Filesize is comparable to Real, but quality is
usually less.
•
Codec has poor cross-platform capabilities (PC, Mac is legacy supported, Windows Mobile support)
•
Multi-bitrate streaming is handled by Microsoft’s proprietary ‘Intellistream’ (or ‘Intelligent Streaming’). This
doesn’t work well on older Windows Media Player clients (less than version 9, we are on version 10 at time of
writing).
•
DRM is possible with Helix DNA DRM and Windows Media DRM / Windows Media Janus. Windows Media
DRM/Janus is the codec’s native DRM technique. Even without DRM applied, streams are somewhat difficult
for the average consumer to record – requiring a specialist piece of software (not common) or intimate
knowledge of their computer’s Sound Card configuration.
•
WMA has poor coverage of codecs shipped in common software players (see list in Appendix A) as cross
platform players don’t have codecs in their versions on non-Windows based machines, Quicktime has a plug-
in available for it to play back WMA.
•
WMA works on 4/15 of the embedded devices and other systems covered in our tests (see list in Appendix A).
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Potential of codec: Windows Media Audio will only ever work properly on Windows Operating Systems from Microsoft,
support on Mac is difficult and it’s a proprietary third-party (i.e. not Microsoft) that now supports a codec
implementation that doesn’t support all of the WMA specifications. It will never work smoothly on Linux, partly because
of its proprietary nature and partly because of the Linux community hating it and refusing to support it. However
Windows Media DRM is used by many Music Stores because Microsoft understands what it is ‘to BETAMAX’ – and
making the licensing of Windows Media DRM fairly open to the industry, compared with Real’s Helix DNA DRM which
requires significant investment to use. However, the continued shipping of Windows Media Player on the majority of
office-based and home PCs keeps this as an important codec which we are entirely right to support in our ‘dual format’
strategy.
Adaptive Multi-rate Family (AMR)
Includes: Adaptive Multi-rate (AMR Narrowband) / Adaptive Multi-rate Wideband (AMR-WB) / Adaptive Multi-rate
Wideband Plus (AMR-WB+, AMR-WBPlus)
The Adaptive Multi-rate Family was initially destined to be covered in this document as it is part of the 3GPP
specification (a mobile handset manufacturer’s forum for standards). This audio codec family is designed to handle
Voice/Speech frequencies only. The family is split into the following codecs:
•
AMR Narrowband – 12.2kbps or less, optimised for Speech only
•
AMR-WB – 24kbps or less, optimised for Speech only
•
AMR-WB+ – 48kbps or less, optimised for Speech only
Characteristics of this codec:
•
Codec has poor cross-platform capabilities (difficult to get the information, but we think mainly on 3GPP
supporting devices)
•
DRM is probable with Helix DNA DRM, but there are no documented uses. It does not have any native DRM
technique.
•
AMR has terrible coverage of codecs shipped in common software players (see list in Appendix A), plug-ins
were not available for most other players we tested. This is not unexpected as the codec is designed for voice
only on mobile handsets.
•
AMR works on maybe 2/15 of the embedded devices and other systems covered in our tests (see list in
Appendix A). This is mainly because it’s part of the 3GPP specification.
Potential of codec: the codec is designed for lower bitrates and for Voice only; it’s also only really supported properly
on 3GPP enabled mobile handsets, so it’s really unlikely to make any impact on the Internet Streaming community.
Summary
We found that the key comparison areas for the codecs were on the number of software players (e.g. iTunes,
RealPlayer) that supported playback of audio files encoded with particular codecs, balanced against the potential
pressure of manufacturers/supporters of common, and upcoming, systems (e.g. WiFi Radio, Mobile handsets, etc) to
use those codecs – resulting in an higher potential for that codec.
We also found that, although there is much technical support for MP3, the codec is rather outdated compared with
newer codecs which have a better filesize to audio-quality balance.
Please see our Conclusions for a list of recommendations based on these findings.
BBC A&M / DISTTECH ALAN OGILVIE Version C.3 13/25
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Conclusions
In an ever changing world, we believe that the BBC’s existing ‘dual format’ minimum of Windows Media and Real
Media is still valid – covering our audience on PC and Mac platforms with a variety of common players (Windows
Media Player, Real Player, VLC, Winamp, etc) and on open source platforms like Linux using VLC.
We believe that it not necessary to apply Digital Rights Management (DRM) to any of our streams (Live or On
Demand). If we were to apply DRM to our streams we would immediately reduce the possible audience, as doing so
would reduce the number of player systems that our streams would work on.
We believe that the most promising audio codec format is AACPlus (HE-AAC), because of improved quality
balanced against reduced bandwidth requirements and the adoption of this format for DAB, Digital Radio Mondiale and
the adoption of it by other Internet Radio stations due to lower bandwidth costs:
•
We recommend adopting AACPlus as a third format for Multicast (i.e. offering Windows Media, Real Media
and AACPlus)
•
We recommend running a trial of an additional Unicast stream using AACPlus
•
We also recommend that we need to form stronger relationships with iTunes/Quicktime (Apple), Windows
Media Player (Microsoft) and Real Player (Real Networks) and Adobe (Flash Player) to make sure that their road
maps include full support for AACPlus.
We believe that MP3 streaming is an old technique, using a codec which is less than efficient (quality vs. bandwidth).
Decent quality MP3 requires higher bit-rate streams to begin with. However, any request for doing MP3 streaming
should be looked at in terms of the audience levels anticipated. Running an MP3 stream for a small number of listeners
would, in theory, be reasonable – however this would have to be balanced against the cost of provision. At the moment
our streaming services aren’t fully scalable, so adding an extra streaming format is a significant cost – if this cost isn’t
outweighed by the potential audience then it’s not sensible to implement.
In covering issues surrounding Audio Codecs, we were asked about Geographical restriction of streams and Rights
Protection. We have two conclusions about this:
•
We recommend a paper which investigates Server Applications that deliver audio in any of the codecs we
have covered, and produce a paper which includes coverage of their implementations for restricting access to
streams based on the geographical location of the listener.
•
We recommend a paper which reviews common Digital Rights Management technologies, and their
impact on audience listening and any restrictions placed on use of audio codecs
BBC A&M / DISTTECH ALAN OGILVIE Version C.2 14/25
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Appendix A:
Codecs matrix
[attached]
BBC A&M / DISTTECH ALAN OGILVIE Version C.3 15/25
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[Attachment for Appendix A - 'Audio Codecs for Internet Streaming'] Report
Audio Codecs
AKA
Quality vs Filesize (Bandwidth)
Noteable Characteristics
Digital Rights Management?
Players with…
...Codecs embedded by default:
Windows Media DRM
Helix DNA DRM - Playback iTunes Fairplay
Does Codec have a native iTunes / QuickTime
VLC (PC/MAC)
only in Real Player / Helix
DRM technique/system?
(PC/MAC)
Client with Helix DNA
Lossy Compression - General
Advanced Audio Coding
AAC
Excellent quality, similar filesize to MP3 but significantly
Multi-bitrate not possible.
better quality in the Audio at the same bitrates. Real uses
AAC for Audio encoded at anything over 96kbps, hence
the support on Real/Helix.
AACPlus
AACPlus,
Excellent quality with lower filesize than AAC, better than Multi-bitrate not possible.
AAC+, High
Real or Windows at circa 56K bitrates, and only small
Efficiency AAC increase required to 64K to end up with very good audio
(HE-AAC)
(not perfect… bit more needed for that - say 96K)
(support likely in
next version)
MPEG audio layer-3
MP3
Compression worst out of others compared on this list.
Multi-bitrate stream doesn't seem possible. Though some
MP3 needs high bitrates to come close to AAC quality at mention of something on Shoutcast documentation but
lower bitrates. MP3 technically requires a license from
difficult to interpret as it doesn't talk about supporting clients
Fraunhoffer even for decoders, though there are some
for playing it back. This is why I don't think it's actually
[less efficient] 'open source' solutions.
possible in Surestream/Intellistream terms.
Real (Real Networks)
RA
Very good at all bitrates - low bitrates are excellent quality Multi-bitrate stream is handled by 'Surestream' and works
for filesize, codec quality better than Windows Media.
quite well.
Rates higher than 96kbps codec switches from Real
(Real Audio plays in Real
native to Real AAC (compatible with standard AAC)
(only very old Real
Player which contains the
codecs supported)
Helix DNA DRM solution)
Vorbis
Ogg Vorbis,
Very good at all bitrates - low bitrates are excellent quality Multi-bitrate not possible.
XIPH Vorbis,
for filesize though filesize is larger than WM/RA, codec
Ogg
quality better than Windows Media. Originally destined to
be a losless codec, Vorbis decided to to for lossy at the
lower bit rates (<96k) but still sounds good. Overall sound
not as good as AACPlus.
WMA (Windows Media Audio)
Low bitrate is problematic. However it redeems itself at
Multi-bitrate stream is handled by 'Intellistream' which
higher bitrates. Filesize is comparable to Real, but quality doesn't work well on older Windows Media Player clients
is usually less.
(less than version 9).
(Windows Media Audio
protected with Windows
Media DRM will play natively
(WMA-3 version of
on PC's with Windows
codec doesn't
Media Player and PC's with
work on Mac)
Real Player where Windows
Media components are
installed)
Lossy Compression - Voice
Adaptive Multi-rate
AMR
Difficult to compare as can't encode or find AMR files.
for bitrates ~12.2kbps or less, designed for Speech
(Narrowband)
Adaptive Multi-rate Wideband
AMR-WB
Difficult to compare as can't encode or find AMR files.
for bitrates ~24kbps or less, designed for Speech
UNKNOWN
UNKNOWN
(but doubtful)
(but doubtful)
Adaptive Multi-rate Wideband
AMR-WB+,
Difficult to compare as can't encode or find AMR files.
for bitrates ~48kbps or less, designed for Speech
Plus
AMR-WBPlus
UNKNOWN
UNKNOWN
(but doubtful)
(but doubtful)
Other Devices and Systems…
...Codecs added by plugin, library or framework:
Winamp
Windows Media
Helix Client
Real Player (PC/MAC)
Flash Player 9+
iTunes / QuickTime
VLC (PC/MAC)
Winamp (PC)
Windows Media Player
Helix Client (PC/MAC)
Real Player (PC/MAC)
Playstation 2
Playstation 2 (Code PSP
Playstation 3
(PC)
Player (PC only)
(PC/MAC)
(PC/MAC)
(PC/MAC)
(PC only)
(Demonstrable)
libraries possible?)
(Apprently - leaked specs
(support for this
-
-
-
(Plugin available - 3ivx
-
-
include support for H.264
wouldn't be surprising
(Apprently in Firmware
Plugin or Orban/Coding
(Likely)
video which is usually
given Adobe and
Version 2)
Technologies Plugin)
found to have AAC
Sorenson collaboration)
audio)
(bug in current released
(support for this
-
-
(Likely if AAC supported,
(Likely if AAC supported,
(Plugin available -
-
-
Windows/Mac version,
wouldn't be surprising
but possibly not full
but possibly not full
Orban/Coding
(Likely)
likely to be fixed by new
given Adobe and
AACPlus sound quality -
AACPlus sound quality -
Technologies Plugin)
Helix Client release)
Sorenson collaboration)
fallback to AAC)
fallback to AAC)
-
-
-
-
-
-
-
(Leaked specs)
-
(Plugin available but
(Plugin available - Real
-
-
(Unlikely due to
(Unlikely due to
(Unlikely due to
need Real Player
Alternative - old codec
proprietary codec - treat
proprietary codec - treat
proprietary codec - treat
components installed
support)
PS2 as Linux machine)
PSP as Linux machine)
PS3 as Linux machine)
anyway!)
UNKNOWN
(but possible as
UNKNOWN
(Nothing for PC version,
-
-
(Plugin available -
-
(Plugin available - Helix
Framework available -
there are some
(but possible)
DirectShow filters by
Community + XIPH
(Likely)
QTComponents by XIPH
XIPH themselves)
themselves)
references to it)
themselves)
(can only play natively if
(Nothing for PC version,
(actively
-
-
-
-
-
(Unlikely due to
(Unlikely due to
(Unlikely due to
Windows Media Player
Framework available for
looking for
proprietary codec - treat
proprietary codec - treat
proprietary codec - treat
is installed on PC, can't
Mac version - Flip4Mac
development)
PS2 as Linux machine)
PSP as Linux machine)
PS3 as Linux machine)
play at all on Mac)
WMV Components)
UNKNOWN
UNKNOWN
UNKNOWN
UNKNOWN
-
-
-
(but doubtful)
(but doubtful)
(but doubtful)
(but doubtful)
UNKNOWN
UNKNOWN
UNKNOWN
UNKNOWN
UNKNOWN
UNKNOWN
(but doubtful)
(but doubtful)
(but doubtful)
(but doubtful)
(but doubtful)
(but doubtful)
UNKNOWN
(but
UNKNOWN
UNKNOWN
UNKNOWN
UNKNOWN
UNKNOWN probable
(but probable
UNKNOWN
UNKNOWN
UNKNOWN
(but probable
(but probable
UNKNOWN
UNKNOWN
(but probable
(but doubtful)
because of
because of 3GPP
(but doubtful)
(but doubtful)
(but doubtful)
because of 3GPP
because of 3GPP
(but doubtful)
(but doubtful)
because of 3GPP
3GPP
support)
support)
support)
support)
support)
More Info
Nintendo DS
Nintendo WII
Helix Client based
Squeezbox
Sonos
KiSS/Cisco NMAs
DAB
Digital Radio
Second Life (uses
Windows Mobile
3GPP Supporting
NMAs (eg. Reciva)
Mondiale
Quicktime Java
based Mobiles
Mobiles
Components)
http://en.wikipedia.org/wiki/Advanced_Audio_Coding
UNKNOWN
(Not natively - though
(specifications not
(the specs of the media
(internal cache sizes on
Squeezebox transcodes
player say this isn't
the DS make it unlikely to
available at this
AAC from it's server
(Likely to update
supported, but some
support any streaming
time for WII)
component to FLAC
firmware)
documents refer to a
audio codecs)
which plays on
Microsoft AAC)
Squeezebox)
http://en.wikipedia.org/wiki/MPEG-4_Part_3
UNKNOWN
(Likely if AAC supported,
(specifications not
but possibly not full
(internal cache sizes on
AACPlus sound quality -
the DS make it unlikely to
available at this
(New version of Helix
(Likely to update
(Likely to update
(Likely to update
(support likely in next
fallback to AAC -
support any streaming
time for WII)
Client will correct bug)
firmware)
firmware)
firmware)
version)
although 3GPP site has
audio codecs)
references to support.
Not sure.)
http://en.wikipedia.org/wiki/Mp3
UNKNOWN
(specifications not
(internal cache sizes on
the DS make it unlikely to
available at this
support any streaming
time for WII)
audio codecs)
http://en.wikipedia.org/wiki/RealNetworks
UNKNOWN
(specifications not
UNKNOWN
(internal cache sizes on
(Possible support on
(Possible support on
the DS make it unlikely to
available at this
(but doubtful)
future firmware update
future firmware update
support any streaming
time for WII)
but seems unlikely)
but seems unlikely)
audio codecs)
http://en.wikipedia.org/wiki/Vorbis
UNKNOWN
(Not natively - though
(specifications not
UNKNOWN
(internal cache sizes on
Squeezebox transcodes
the DS make it unlikely to
available at this
AAC from it's server
(but possible)
support any streaming
time for WII)
component to FLAC
audio codecs)
which plays on
Squeezebox)
http://en.wikipedia.org/wiki/Windows_Media_Audio
UNKNOWN
(specifications not
(internal cache sizes on
(Interesting: WMA
(Certainly Reciva models
(Possible support on
the DS make it unlikely to
available at this
Losless isn't supported,
(No support for Windows
have licensed WMA
future firmware update
support any streaming
time for WII)
but for this discussion it
Media Lossless)
codecs)
but seems unlikely)
audio codecs)
doesn't matter)
UNKNOWN
http://en.wikipedia.org/wiki/Adaptive_Multi-Rate
UNKNOWN
(but probable
(specifications not
UNKNOWN
UNKNOWN
UNKNOWN
(internal cache sizes on
because of
the DS make it unlikely to
available at this
(but doubtful)
(but doubtful)
(but doubtful)
support any streaming
Quicktime-3GPP
time for WII)
audio codecs)
support)
UNKNOWN
http://en.wikipedia.org/wiki/AMR-WB
UNKNOWN
(but probable
(specifications not
UNKNOWN
UNKNOWN
UNKNOWN
(internal cache sizes on
because of
the DS make it unlikely to
available at this
(but doubtful)
(but doubtful)
(but doubtful)
support any streaming
Quicktime-3GPP
time for WII)
audio codecs)
support)
http://en.wikipedia.org/wiki/AMR-WBplus
UNKNOWN
UNKNOWN
UNKNOWN
(but probable
(specifications not
UNKNOWN
UNKNOWN
UNKNOWN
(but probable
(internal cache sizes on
because of
the DS make it unlikely to
available at this
(but doubtful)
(but doubtful)
(but doubtful)
because of 3GPP
support any streaming
Quicktime-3GPP
time for WII)
support)
audio codecs)
support)
Appendix B:
Codecs not covered
[attached]
BBC A&M / DISTTECH ALAN OGILVIE Version C.3 19/25
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[Attachment for Appendix B - 'Audio Codecs for Internet Streaming'] Report
Audio Codecs
AKA
Similar
Noteable Characteristics
Uncompressed
AIFF
RIFF
WAV
Losless Compression
Apple Lossless
Audio Lossless Coding
MPEG-4 ALS
Direct Stream Transfer
DST
Free Lossless Audio Codec
FLAC
LA
Lossless Audio
Lossless Predictive Audio Compression
Lossless Transform Audio Compression
LTAC
Meridian Lossless Packing
Monkey's Audio
APE
OptimFROG
RealAudio Lossless
RKAU
RK Audio
Shorten
SHN
True Audio
TTA
WavPack
Windows Media Audio 9 Lossless
WMA Lossless
Lossy Compression - General
ADPCM
Whilst supported by many players,
ADPCM isn't of comparable compression
to the others chosen for comparison
ADX
ARDOR
Adaptive Rate-
Distortion
Optimised
sound codeR
ATRAC Adaptive TRansform Acoustic Coding
ATRAC, ATRAC-
3
Dolby Digital (A/52, AC3)
DTS DTS
Coherent
Acoustics
Impala FORscene audio codec
MPEG audio layer-1
MP1 Audio
MPEG audio layer-2
MP2 Audio
HILN
MPEG-4
Parametric
audio coding
Musepack
Perceptual Audio Coding
QDesign
TwinVQ
Lossy Compression - Voice
AMBE
CELP
DSS
EVRC
FS-1015 (LPC-10)
FS-1016 (CELP)
G.711 (a-law and mu-law/u-law)
G.721 (Superseded by G.726)
G.722.1
G.722.2
G.722
G.723.1
G.723 (Superseded by G.726)
G.726 (ADPCM)
G.728 (LD-CELP)
G.729 (CS-ACELP)
G.729a
G.729.1
GSM Enhanced Full Rate
GSM Full Rate
GSM Half Rate
iLBC
IMBE
iSAC
QCELP
Part of the 3GPP specification, but not
used any more - Originally developed by
Qualcomm in 1994 for bitrates < 16kHz
Relaxed Code Excited Linear Prediction (RCELP)
SMV
Speex, patent free
VSELP
Appendix C: Assumptions
Please consider these assumptions whilst reading the Codecs worksheet in Appendix A, or the Codecs section in the
main body of the paper:
•
The listening quality of the codecs is not of primary focus here, but rather the business logic surrounding them.
Consider the tale of VHS versus BETAMAX. Even though BETAMAX was technically better, VHS won out due
to the business factors (more manufacturers embedded VHS systems because it was cheaper than BETAMAX,
though VHS tapes wore out far quicker than BETAMAX and were inferior technology). This is called ‘to
BETAMAX’, where a technology that was technically better lost out due to market forces. (If you are interested
in the full story of BETAMAX see:
http://en.wikipedia.org/wiki/Betamax)
•
We will not be looking at Losless Compression or Uncompressed audio - our task is to examine codecs that
could be used for delivery via Unicast/Multicast streaming, Losless or Uncompressed aren't things that we
should do in this area. Some Lossy Compression voice codecs are looked at because they form part of the
3GPP specifications, 3GPP is supported by many mobile manufacturers.
•
Windows Media Player 10 Mobile, which is embedded on Windows Mobile based devices (e.g. XDA, Orange
SPV, Pocket PC devices, etc) - does not support 3GPP standards for audio, only Windows Audio and
Windows Video, MP3 and ISO MPEG-4 (it doesn't say whether this includes the audio codecs appropriate to
3GPP specs)
http://www.microsoft.com/windows/windowsmedia/player/windowsmobile/faq.aspx#5_2
•
3GPP - because of the number of mobile handsets available it is impossible, without extended study, to review
every single handset. Because of this I have decided to look for the number of major handset manufacturers
that are partners in the 3GPP plan. Sony Ericsson, Nokia, Samsung, LG, Motorola, Siemens/BenQ, NEC,
Sagem
•
3GPP spec contains the following audio codecs - AAC and AMR. I make the assumption that if a Player or
Device supports 3GPP then it supports at least AAC, and possibly AMR.
•
Helix Client - is a multi-format supporting piece of software, released to community development by Real
Networks. The client can be used in Software based players, and Hardware firmware. Helix supports dozens of
audio codecs, a full list and some more information about Helix is available here:
https://datatype.helixcommunity.org/
http://en.wikipedia.org/wiki/Helix_project
•
Networked Media Appliance (NMA) - is any such device that looks like an Internet Radio or Media Server.
E.g. WiFi Internet Radios like Reciva, etc. These appliances tend to have firmware which is upgradeable over
the internet from the manufacturer. Some can upgrade the codecs used.
•
Often the codecs themselves do not dictate whether they are supported by a player - sometimes the
container in which they are placed does. The container specifies the way the encoded data is packetised and
this can lead to problems. Video containers are much more complicated than Audio, so I am assuming that if a
codec is supported by a particular Player or Device then it works and the container in which it is placed is
irrelevant.
•
VLC Media Player is available across a number of platforms, I will only be reviewing codecs against the PC
and Mac versions. Though it is very good for Codec support on Linux, Windows Media and Real formats don't
BBC A&M / DISTTECH ALAN OGILVIE Version C.3 21/25
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really work or work with only older versions of the codecs. See the following for full support information:
http://www.videolan.org/vlc/features.html
•
Flash Video (FLV) codecs for Video use Sorenson Spark video codecs and the audio codec is either MP3 or
Uncompressed. This is why I have not put this in as a codec, as for audio purposes it is the same as MP3.
•
The column marked - 'Quality vs Filesize (Bandwidth)' - is purely a simple, indicative description of what I
gauged opinion to be.
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Appendix D: Current Bit-rates
Windows Media - Live Streams
UNITED KINGDOM
KEY TO SHORTHAND: 32k = 32kbps / 44z = 44kHz / S = Stereo / M = Mono
Network/Sub-Network
Current
Proposed - NOW
Radio 1
-
20k(32z) M,
48k(32z) S
Radio 2
-
20k(32z) M,
48k(32z) S
Radio 3
-
20k(32z) M,
48k(32z) S
Radio 4
-
20k(32z) M,
48k(32z) S
Radio 4 LW
-
20k M, 32k M
Five Live
32k(44z) M
20k M, 32k M
Five Live Sports Extra
32k(44z) M
20k M, 32k M
6Music (Offering for narrowband)
10k(11z) M, 16k(16z) M,
MERGED
20k(32z) M, 32k(44z) S,
40k(32z) S
6Music (Offering for broadband)
48k(32z) S
MERGED
6Music
-
20k M, 48k S
BBC7 (Offering for narrowband)
20k(32z) M, 32k(44z) S,
MERGED
40k(32z) S
BBC7 (Offering for broadband)
48k(44z) S
MERGED
BBC7
20k M, 48k S
1xtra (Offering for narrowband)
20k(32z) M, 32k(44z) S,
MERGED
40k(32z) S
1xtra (Offering for broadband)
20k(22z) S, 32k(44z) S,
MERGED
48k(32z) S
1xtra
20k M, 48k S
Asian Network (Offering for nar-
20k(32z) M, 32k(44z) S,
MERGED
rowband)
40k(32z) S
Asian Network (Offering for
48k(44z) S
MERGED
broadband)
Asian Network
-
20k M, 48k S
Windows Media - AOD Streams
UNITED KINGDOM
KEY TO SHORTHAND: 32k = 32kbps / 44z = 44kHz / S = Stereo / M = Mono
Network/Sub-Network
Current
Radio 1
NONE
Radio 2
NONE
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Radio 3
NONE
Radio 4
NONE
Radio 4 LW
NONE
Five Live
NONE
Five Live Sports Extra
NONE
6Music NONE
BBC7 NONE
1xtra NONE
Asian Network
NONE
Real Media - Live Streams
UNITED KINGDOM
KEY TO SHORTHAND: 32k = 32kbps / 44z = 44kHz / S = Stereo / M = Mono
Network/Sub-Network
Current
Proposed - NOW
Radio 1
11k(5.5z) M, 16k(8z) M, 20k(10z) M,
SAME
32k(13.8z) S, 44k(13.8z) S
Radio 2
11k(5.5z) M, 16k(8z) M, 20k(10z) M,
SAME
32k(13.8z) S, 44k(13.8z) S
Radio 3
11k(5.5z) M, 16k(8z) M, 20k(10z) M,
SAME
32k(13.8z) S, 44k(13.8z) S
Radio 4
8k(4z) M, 16k(8z) M, 32k(8z) S, 44k(11z)
SAME
S
Radio 4 LW
8k(4z) M, 16k(8z) M, 20k(10z) M, 32k(16z)
SAME
M
Five Live
8k(4z) M, 16k(8z) M, 44k(20z) M
SAME
Five Live Sports Extra
8k(4z) M, 16k(8z) M, 44k(20z) M
SAME
6Music
11k(5.5z) M, 16k(8z) M, 20k(10z) M,
SAME
32k(13.8z) S, 44k(13.8z) S
BBC7
11k(5.5z) M, 16k(8z) M, 20k(10z) M,
SAME
32k(13.8z) S, 44k(13.8z) S
1xtra
11k(5.5z) M, 16k(8z) M, 20k(10z) M,
SAME
32k(13.8z) S, 44k(13.8z) S
Asian Network
20k(9.9z) S, 32k(13.8z) S, 44kbpz(16z) S
11k(5.5z) M, 16k(8z)
M, 20k(10z) M,
32k(13.8z) S,
44k(13.8z) S
Real Media - AOD Streams
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UNITED KING-
KEY TO SHORTHAND: 32k = 32kbps / 44z = 44kHz / S = Stereo / M =
DOM
Mono
Network/Sub-Network
Current
Proposed - NOW
Radio 1
SAME AS LIVE
SAME AS LIVE
Radio 2
SAME AS LIVE
SAME AS LIVE
Radio 3
SAME AS LIVE
SAME AS LIVE
Radio 4
SAME AS LIVE
SAME AS LIVE
Radio 4 LW
SAME AS LIVE
SAME AS LIVE
Five Live
SAME AS LIVE
SAME AS LIVE
Five Live Sports Extra
SAME AS LIVE
SAME AS LIVE
6Music
SAME AS LIVE
SAME AS LIVE
BBC7
SAME AS LIVE
SAME AS LIVE
1xtra
SAME AS LIVE
SAME AS LIVE
Asian Network
SAME AS LIVE
SAME AS LIVE
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