Articles & Transcripts


Remarks on Mastering Over the Years (from Bob Ludwig, 2015)

Bob got his masters degree from the Eastman School of Music, and started his audio engineering career there by cutting lacquers for vinyl LP’s.  His discerning ear is certainly at least in some part due to his training as a musician (he played trumpet with the Utica Symphony Orchestra).

Bob created the reference lacquer for The Band, “Music from Big Pink”.  He tweaked cutting the disc in order to try and get more low bass on it.  However, the bass was subsequently filtered out on the commercial pressing - probably in order to prevent skipping on cheaper turntable/stylus setups that would be unable to track the grooves.

Bob did the lacquer for the Frank Zappa album, “Sheik Yer Booty”.  There were a large number of tweaks including EQ on individual words, and filtering for certain passages; he states he had to memorize parts of the music in order to get all his modifications done correctly.

Bob did the lacquer from the master of Steely Dan’s record, “Gaucho”, which was mixed to 15” tape using Dolby A noise reduction (mixed by Elliot Scheiner).  This album utilized a drum machine built by the bands engineer, Roger Nichols (the bands recording engineer).  Bob loved the mix and the record.  Bob also did some mastering for Donald Fagen’s solo album, “The Nightly”.  By that time Roger Nichols was using an early Sony digital system for recording, but Bob says it still sounded great (despite the limitations of that early digital technology).

Bob re-edited/remastered the entire Rolling Stones catalog for Virgin records in the 1990’s; some feel these are still the best remasters to date.  The album “Tattoo You” was assembled mostly from outtakes of previous recording sessions.  Apparently this was the last Stones album that reached the top of the charts.  It was also a long record, and it was hard to fit onto a vinyl pressing.


Audio and MP3 (SV&C article,  by SynMuse Productions)

A look (back) at the mechanisms that drive MPEG-compressed audio and some of the ways that the contractor can apply this increasingly accepted medium.


MPEG(Moving Picture Experts Group) audio and video have been with us now for more than 10 years. The MPEG group was formed in the late 1980s to create standards for the compression of digital audio and video signals. In 1992, MPEG became a standard as agreed upon by the International Standards Organization (ISO) and the International Electrotechnical Commission (IEC). MPEG 2 became a standard in 1994 and added the ability to encode content at lower bit rates (16 kbps, 22.05 kbps, 24 kbps) and to encode a signal according to psychoacoustic models. The psychological models exploit masking and threshold effects in human hearing to decrease the amount of encoded data below the audibility threshold. For video encoding, MPEG may decrease the number of bits per frame for less complex pictures because the human eye will not notice the loss of quality. There are three layers to the MPEG specification that apply to audio (only) encoding—layer one, layer two, and the most recent, layer three.

Interestingly, the most prolific users of MP3 are musicians and Web surfers who have all but created an online culture around the exchange of MP3 (music only) files, so most of the information gleaned about this cutting-edge technology has been through its application on the Internet. Although it may be difficult in the infancy of a new trend to determine its ultimate effects on the A-V world, what is clear is that MP3 is here to stay, and it is evolving. The format of this transmission, compression and decoding scheme will no doubt undergo many changes over time to suit the needs of its users and exploit what will be an ever-changing computer hardware marketplace.

So what does this technology mean to the installation and contracting professional? MPEG offers a unique way of transmitting and receiving audio and video data over the Internet. In the future, manufacturers will be able to design devices that can be updated with such new program material as music and messages as well as program material that is developed, modified and mixed via computer. An advantage to this technology is its media—it does not wear out like tape or other magnetic media, nor does it require more permanent media to be produced, such as CD-ROM or DVD. It can be continuously updated via an Internet connection, thereby avoiding costly on-site service. Venues where extremely high-fidelity program material is not paramount, such as background music applications or video in such environments as restaurants, bars and retail applications, are likely to benefit the most at a point in time sooner than other venues as the technology develops further.

Although MPEG 2 delivers audio and picture quality equivalent to TV studio standards, it is not perfect. MPEG is not a loss-free encoding or compression scheme, and its application results in a loss in signal quality. A way to understand compression loss lies in understanding that you do not get 100% signal quality after compression, but by setting the encoding parameters in advance of the compression so that you can ensure that the material you encode and subsequently transmit, download or read from a CD is of a high enough quality for your target audience. Another point to remember is that the encoding setups will differ with different applications, using different schemes for broadcast, downloading over the Internet, or producing a CD.

Defining the technology

A single piece of hardware (or software) that can do both encoding and decoding is sometimes referred to as a codec (encoder/decoder). In audio, the current MPEG specifications are broken up into layers, termed 1, 2, and 3. The popular abbreviation, MP3, refers to audio layer three encoding. No one seems to know why 128 kbps MP3 became the choice for downloading files from the Internet instead of 128 kbps MP2. In all likelihood, it happened this way because MP3 is a more recent development than MP2, and although MP3 is a higher revision version than MP2, people sometimes assume MP3 is superior. It is a fact that MP3’s predecessor was audio layer two or MP2 encoding, and many people believe it to be superior to MP3 at bit rates of 128 kbps or higher.

Higher fidelity encoding, however, requires more resources, and this means more bandwidth and an increased demand for data storage space. Higher layers increase the amount of audio data compression and the complexity of encoding the audio signal. It is less mathematically intensive (and therefore takes less time) to encode a signal on layer one as audio than it is to encode the same signal on layer two as audio. The layers are hierarchical, so a layer three decoder should be able to decode layer two audio. Layer three is built on the features of layer two, adding a modified FFT (Fast Fourier Transform) and a modified discrete cosine transform to the encoding process. Encoding for layer three is more computationally intensive then layer two or layer one. The more complex encoding schemes and algorithms of the higher layers can improve audio quality, despite having greater compression. Even with increased compression and a lower bit rate, layer three audio encoding offers equal or greater quality then layer two audio encoding. For the overall effectiveness of MPEG audio’s different layers, refer to table 1.

Economy of scale: time vs. audio quality

Increased computation time on the encoding side is a small price to pay for the quality and compression that even MP3 affords. Thus, MP3 encoding is starting to be applied at even the professional audio level. For example, 4 minutes of audio from a standard audio CD requires about 40 MB of disk or server space. The equivalent MP3 or MP2 file encoded at a 128 kbps constant bit rate takes up about 4 MB of space, a tenth of the space (a 10:1 compression ratio).

Some audiophiles describe the quality of MP3 audio at 128 kbps as not being even remotely close to CD. Most people, however, hear 128 kbps constant bit rate MP3 audio as comparable to a Dolby B or Dolby C cassette recording of a state-of-the art CD; there is a reduction in the dynamic range and some loss of highs and imaging, but content will remain a far cry from unlistenable. Different codecs can provide varying levels of audio quality, and more importantly, such encoding parameters as the encoding model or algorithm, where to cutoff low frequencies, and the choice of stereo modes can affect the sound quality any MP3 encoder will produce. Decoders can vary in quality in similar ways.

Tradeoffs: bit rate vs. bandwidth

An important consideration in encoding audio is the relationship between audio quality and bit rate (or bandwidth) and how much space the data requires on disk or in memory. If you encode at lower bit rates, audio quality can suffer, but lower bit rates are better suited to slower speed network and transmission lines. Similarly, files encoded at lower bit rates also take up less size in memory or to data storage. If you are willing to double your bandwidth from 128 kbps to 256 kbps, then constant bit rate MP2 or MP3 audio is fairly close and perhaps indistinguishable from CD quality. The 4 minute selection example mentioned earlier now requires about 8 MB of disk space when encoded in 256 kbps constant bit rate MP2 audio, or you get a 5:1 compression ratio.

Further, doubling the bandwidth from 128 kbps to 256 kbps to increase the audio quality halves the compression ratio from 10:1 to 5:1 and doubles the storage to contain the entire file all at once on disk or in memory. Broadcasters will also need to rent or buy faster network connections to transmit audio at higher bit rates.

Downloading

Downloading audio means receiving audio data from a server over the network. Downloads of files usually require the entire file to be copied to disk before anything can subsequently be done with the file (like playing it). Therefore, downloading files of MP3 music means the end user waits for the entire file to be copied over the network to his local disk before playing it. If you are transmitting data to a client with a 56 kbps modem, the 4 minute, 4 MB MP3 file will take about 10 minutes to download, assuming the network connection between your computer and the server does not encounter severe degradation or bottlenecks. The 8 MB MP2 file would take twice as long, or about 20 minutes to download, but compare this to the amount of time it would take to download the original 40 MB CD audio file—100 minutes.

Downloading music files in their entirety is an expensive operation in terms of time, and as such, it is not great for broadcasting or real-time applications. If the download gets interrupted by a network or transmission line failure, you usually have to start again, unless some rather smart network protocols are employed. However time consuming and tedious, an advantage of downloading is that you only have to do this operation once, and then you have your own private copy of the music or other data on your local disk. If there are no copy protections on this file, you can duplicate it as many times as you like. With such programs as WinAmp for PC users and MacAmp for Apple users, many people on the Web are beginning to collect MP3 files on their computers and trade them with others. These JukeBox programs create playlists, and you can organize your music files allowing one to program their playback in any order or fashion. If a song in the form of an MP3 file gets popular, it can be copied and transmitted over the network among fans hundreds of times in a day or two. Musicians, of course, love this. Record labels, however, typically loathe this practice.

Streaming

Streaming audio is the ability to start playing audio before it has been downloaded into your system from the Web as a complete file. This is necessary because of the time needed for a complete download, and it allows the listener to have access to the material much more quickly in the process. By buffering and assembling the bits as they are received, an MP3 decoder can start to play audio almost right away. The stream is played in real-time, and a copy of the entire file need not be assembled and saved to your local disk.

For example, a player with a buffering scheme that stores up to 30 seconds of music might start to play music after it has downloaded only the first 5 seconds of music from the Internet. The 5 second or so time lag between receiving and playing audio is a small price to pay for the improved real-time performance. Also, decreasing the playback bit rate to something less then 56 kbps (like 28 kbps) ensures that there is a steady stream of music; the player will not run out of music to play before enough new music is downloaded to and buffered in the player. Streaming is really a clever tradeoff that delays playing music in real-time but is not so costly in terms of time as waiting for the music to download in its entirety.

Unlike downloading entire files that provide a complete copy of the music on your hard disk, the piecemeal technique of streaming is sensitive to problems with the network and transmission lines. If the network gets interrupted for longer than the player can buffer music, then the stream of music will be broken and the player will produce an audible skip, which sounds like static or background noise. Because most people are wired to the Internet over consumer-grade phone lines, they will inevitably experience bit rates of much less than even 53 kbps from network congestion and bottlenecks. Streaming is going to skip sometimes as a result, and it is going to take even longer to download the complete file or broadcast. High-end users with cable modems, ISDN and ADSL may still encounter bottlenecks downloading data from a server, but they can generally stream audio at higher bit rates. When streaming audio, we are normally constrained by the bit rates available over conventional telephone networks, and audio quality suffers. It is not yet a perfectly networked world.

Let me add a quick note about codecs. Codec manufacturers are still developing their algorithms. It may not be surprising to find that algorithms that sound good for encoding speech can actually sound lousy when encoding music. What is surprising is that algorithms designed to extend the high end for encoding broadcast music in MP3 often do not sound good for speech range material. The best way to find out which algorithms work best for your program material is to audition your program material with the different encoding schemes available on your codec.

Legal issues

No one paid much attention to the legal implications MP3 files bring to use of commercial music on the Web until publicity broke about a Stanford University sophmore who had posted his collection of favorite music on a university server. The server was taking so many hits that it began to attract attention. University networks are in no way immune from the law, but what occurs behind private university or corporate network firewalls is unlikely to be scrutinized heavily from a legal standpoint despite official regulations, and therein lies the copyrighting issue that owners of material being posted or moved around on the Web most fear.

It is probably inappropriate for this article to describe at length all the legal ramifications of posting commercial music or intellectual property on the internet, but some guidelines are in order for streaming or downloaded audio. There is nothing illegal about the copyright holder posting MPEG audio files of his or her work on the network, and anyone can subsequently copy those files as many times as they like or propagate them anywhere on the network. The music industry claims that this is not what concerns the authorities, and they mostly dismiss this to be a fringe market populated by musicians seeking publicity.

The major recording labels claim that they do care about anyone’s duplicating or ripping off the music or intellectual property of their artists for the purpose of making it freely available on the Web because they receive no royalty payment. Although the copyright law allows you to make a reasonable number of copies for your private use, this concept of reasonable use does not include posting your favorite copy-protected songs on your home page at AOL as far as the record labels are concerned.

Copyright ID and protection

MPEG audio can be encoded with different kinds of ID tags to identify such things as the copyright owner, song title, artist and album. Unfortunately, these tags do not provide any kind of physical copy protection. The music industry is literally clamoring to provide its own secure electronic music distribution scheme known as SDMI (Secure Digital Music Initiative).

Like schemes before it, SDMI provides a digital watermarking scheme that imbeds a virtually inaudible digital signature in the file as well as a copy protection scheme. If everyone uses an SDMI-compliant player, then SDMI watermarked files could be played, and their distribution could be tracked and royalties collected. Once the copy count was exceeded, playing or copying the file would not be possible. Although SDMI has an impressive number of companies as members, there is no reason to believe, based upon Internet culture, that SDMI will replace free MP3 (and MP4 in the offing) as the de facto standard for internet audio. Manufacturers are free to make SDMI-compliant players that do not exclude playing non-SDMI encoded files.

History has shown that wherever intellectual property is distributed, regulation of intellectual property law is sure to follow. First, there were printing rights, then came audio and video rights. Now, there will be Internet rights and portable-device rights. Welcome to the brave new future of electronically transferred audio.

View the Figures

For more information on compression schemes, visit www.cs.sfu.ca/undergrad/CourseMaterials/CMPT479/material/notes/Chap4.

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