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    Digital Audio Format Guide "which is best for you?"

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    Digital Audio Format Guide "which is best for you?"

    Post by maxim9691 on Mon Jan 25, 2010 12:57 pm

    Compression Formats: Quantity vs. Quality

    First the beginning of the Compression Format for music
    MP3, which stands for MPEG Audio Layer 3, is a music audio format that has revolutionized the music world by starting a whole new culture, especially on the web. Every music lover is aware of the word ‘MP3’, but very few know the whys and hows of it.
    How it all began - Karlheinz Brandenburg and Dieter Seitzer

    It all began way back in 1977, when Karlheinz Brandenburg, a specialist in mathematics and electronics in Germany, started researching methods of compressing music. At the same time Prof. Dieter Seitzer of Erlangen-Nuremberg University, Germany, also began his struggle of trying to compress music over phone lines. It was the advancement of digital technology that had made them think of compressing music. They knew that existent music was almost impossible to send over phone lines. So, they wanted to produce a music file that was much smaller and yet sound the same as the original uncompressed audio.
    Karlheinz Brandenburg and Dieter Seitzer continued working independently till 1987, when the Fraunhofer Institut in Germany undertook the research project under code-name EUREKA project EU147, Digital Audio Broadcasting (DAB). Both Karlheinz Brandenburg and Dieter Seitzer joined the project along with Bernhard Grill, Thomas Sporer, Bernd Kurten, and Ernst Eberlein. In 1989, the MP3 technology was finally developed and its patent (German) was obtained.  Meanwhile, in 1988, an international working group of ISO/IEC charged with the development of video and audio encoding standards, called the Motion Pictures Experts Group, or MPEG, was formed. MPEG soon started testing the various audio codecs submitted to it for approval. The ones that excelled in the testing were taken and further refined and integrated into one codec. It was here that Karlheinz Brandenburg’s and Dieter Seitzer’s audio coding algorithm was used as the basis and integrated into MPEG’s new audio codec. The new codec was released in 1992 as MPEG-1. The standard was later developed into MPEG-2 and published in 1994. The third version, MPEG Audio Layer 3, or MP3 was released on November 26, 1996, and the United States patent was issued. Fraunhofer Institut soon began asking for patent rights, and in September 1998 finally won. Now all developers of MP3 encoders/rippers and decoders/players have to pay a licensing fee to Fraunhofer.

    First Music In MP3

    In February 1999, SubPop became the first company to produce and distribute music tracks in the MP3 format. Soon after, the portable MP3 player appeared. The popularity of MP3 was now fast rising, especially on the internet. Time saw the music industry completely changed. No longer were music lovers rushing to stores as music could now be easily downloaded from the Internet or "ripped" from a music CD. Neither was there need to search for an audio CD that one wanted to hear, because now it was possible to store thousands of songs on a small portable player and then search by title, artist, album or genre. Whether at home or outside, all that had to be done to listen to a song was press a button.

    MP3 Players

    The MP3 Player also progressed with time. The very first commercially released Player was the MPMan F10, launched by Korea’s Saehan Information Systems in 1998. It was a 91 x 70 x 165.5 mm device that contained 32MB of Flash storage, and had to be connected to a parallel port on a PC to copy songs. Later the same year, another player, the Diamond Multimedia Rio PMP300 was released in the U.S. This, too, had 32MB of storage and had to be connected to a parallel port. The only difference was that it contained a Smart Media slot that allowed users to increase storage space. Despite the release and wide use of these MP3 players, the explosive growth came in 2003, when Apple allowed Windows PC owners to use the iPod, their new MP3 device. The initial iPod, released in 2001, was meant only for Macintosh users. It was the first hard-drive based device with 5 GB of space. Today there is a wide range of devices produced by hundreds of software companies the world over that users can select from, depending on their requirements.  These players may be categorized on the basis of their storage media – flash based, hard drive based, MP3 CD, and hybrid. The most popular players today (according to CNET) are Apple iPod, Microsoft Zune, Archos WiFi, Creative Zen, and iRiver Clix.

    Limitations Of MP3 Give Birth To Other Audio Formats

    Despite its unprecedented popularity, MP3 does have its limitations. Among them – time resolution is too low, it often creates echoes, coding efficiency is not very good due to double filter banks, no scale factor bands for high frequencies, and encoding delays due to internal handling of bit reservoir. Due to these deficiencies several new audio formats were developed. The most poplar of these alternatives to MP3 are AAC and WMA. These are just as widely supported as MP3 and do not have the limitations that MP3 has. Other audio formats that though are not as widely supported but are still popular because they are open sourced and completely free, are Ogg Vorbis, FLAC, and Speex.

    Regardless of everything, MP3 continues to be the dominant music audio format and the favorite of music lovers. The reasons for this are simple – it is free, is easy to make, requires a much smaller saving space, and most music software include options to create or convert files to MP3. Today there are thousands of websites that offer MP3 music files for "censored", and millions of music lovers from all over the world are downloading music daily. Many sites are illegal, so in order to avoid risks and safety concerns, it is important to go for legal ones which have been sanctioned by the music industry. The most popular of these are Rhapsody, Napster, iTunes, eMusic, Amazon, and Epitonic.
    +MP3 In Music Industry Today+
    This brings us to another point – the impact MP3 has had on this segment of the music industry, i.e. the distribution segment. The popularity of the internet and the rise in online music distribution has facilitated the entry of small firms into the field. In other words, barriers to entry have eased. The need for promotional activities has also lessened. This is because the music distribution websites allow consumers to view song clips thus allowing them to actually see/hear what they are buying. Production costs of producers have also been cut because of the downloading - not as many CDs have to be produced. Thus both the consumer and the producer have benefited from online music distribution – which was a result of MP3.      
    It will not be wrong to say that MP3 has revolutionized the music industry. Since the time it has appeared, it has become a cultural phenomenon, something much more than mere technology. It has brought about a new era in the music industry.
    Above Article by Olga Lapshinova

    MP3. The "Industry Standard". Just about every player supports this format. This is often what you'll find if "censored" music online, and gives you the most options in terms of encoders and playback software on your computer. Some say that MP3 has been outpaced by newer, better sounding formats such as WMA, AAC, and OGG. If you choose this format, try and use "Variable Bit Rate" MP3 encoding. This is a smarter method of encoding which consumes more drive space on more complex parts of a musical composition, where there's more information, and less in quieter or less complex passages.
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    Windows Media Audio (WMA)

    Post by maxim9691 on Mon Jan 25, 2010 1:06 pm

    Windows Media Audio (WMA) is an audio data compression technology developed by Microsoft. The name can be used to refer to its audio file format or its audio codecs. It is a proprietary technology that forms part of the Windows Media framework. WMA consists of four distinct codecs. The original WMA codec, known simply as WMA, was conceived as a competitor to the popular MP3 and RealAudio codecs.WMA Pro, a newer and more advanced codec, supports multichannel and high resolution audio. A lossless codec, WMA Lossless, compresses audio data without loss of audio fidelity. And WMA Voice, targeted at voice content, applies compression using a range of low bit rates.


    Development history

    The first WMA codec was based on earlier work by Henrique Malvar and his team which was transferred to the Windows Media team at Microsoft. Malvar was a senior researcher and manager of the Signal Processing Group at Microsoft Research,whose team worked on the MSAudio project.The first finalized codec was initially referred to as MSAudio 4.0. It was later officially released as Windows Media Audio, as part of Windows Media Technologies 4.0. Microsoft claimed that WMA could produce files that were half the size of equivalent-quality MP3 files; Microsoft also claimed that WMA delivered "near CD-quality" audio at 64 kbit/s. The former claim however was rejected by some audiophiles.RealNetworks also challenged Microsoft's claims regarding WMA's superior audio quality compared to RealAudio.

    Newer versions of WMA became available: Windows Media Audio 2 in 1999,Windows Media Audio 7 in 2000,Windows Media Audio 8 in 2001, and Windows Media Audio 9 in 2003 Microsoft first announced its plans to license WMA technology to third-parties in 1999. Although earlier versions of Windows Media Player played WMA files, support for WMA file creation was not added until the seventh version. In 2003, Microsoft released new audio codecs that were not compatible with the original WMA codec. These codecs were Windows Media Audio 9 Professional,Windows Media Audio 9 Lossless, and Windows Media Audio 9 Voice.
    Container format
    A WMA file is in most circumstances encapsulated, or contained, in the Advanced Systems Format (ASF) container format,featuring a single audio track in one of following codecs: WMA, WMA Pro, WMA Lossless, or WMA Voice. These codecs are technically distinct and mutually incompatible. The ASF container format specifies how metadata about the file is to be encoded, similar to the ID3 tags used by MP3 files. Metadata may include song name, track number, artist name, and also audio normalization values.

    This container can optionally support digital rights management (DRM) using a combination of elliptic curve cryptography key exchange, DES block cipher, a custom block cipher, RC4 stream cipher and the SHA-1 hashing function. See Windows Media DRM for further information.
    Codecs
    Windows Media Audio

    Windows Media Audio (WMA) is the most common codec of the four WMA codecs. Colloquial usage of the term WMA, especially in marketing materials and device specifications, usually refers to this codec only. The first version of the codec released in 1999 is regarded as WMA 1. In the same year, the bit stream syntax, or compression algorithm, was altered in minor ways and became WMA 2. Since then, newer versions of the codec were released, but the decoding process remained the same, ensuring compatibility between codec versions. WMA is a lossy audio codec based on the study of psychoacoustics. Audio signals that are deemed to be imperceptible to the human ear are encoded with reduced resolution during the compression process.

    WMA can encode audio signals sampled at up to 48 kHz with up to two discrete channels (stereo). WMA 9 introduced variable bit rate (VBR) and average bit rate (ABR) coding techniques into the MS encoder although both were technically supported by the original format,. WMA 9.1 also added support for low-delay audio, which reduces latency for encoding and decoding.

    Fundamentally, WMA is a transform coder based on modified discrete cosine transform (MDCT), somewhat similar to AAC, Cook and Vorbis. The bit stream of WMA is composed of superframes, each containing 1 or more frames of 2048 samples. If the bit reservoir is not used, a frame is equal to a superframe. Each frame contains a number of blocks, which are 128, 256, 512, 1024, or 2048 samples long after being transformed into the frequency domain via the MDCT. In the frequency domain, masking for the transformed samples is determined, and then used to requantize the samples. Finally, the floating point samples are decomposed into coefficient and exponent parts and independently huffman coded. Stereo information is typically mid/side coded. At low bit rates, line spectral pairs (typically less than 17 kbit/s) and a form of noise coding (typically less than 33 kbit/s) can also be used to improve quality.

    Like AAC and Ogg Vorbis, WMA was intended to address perceived deficiencies in the MP3 standard. Given their common design goals, it's not surprising that the three formats ended up making similar design choices. All three are pure transform codecs. Furthermore the MDCT implementation used in WMA is essentially a superset of those used in Ogg and AAC such that WMA iMDCT and windowing routines can be used to decode AAC and Ogg Vorbis almost unmodified. However, quantization and stereo coding is handled differently in each codec. The primary distinguishing trait of the WMA Standard format is its unique use of 5 different block sizes, compared to MP3, AAC, and Ogg Vorbis which each restrict files to just two sizes. WMA Pro extends this by adding a 6th block size used at 88.1/96 kHz sampling rate.

    Certified PlaysForSure devices, as well as a large number of uncertified devices, ranging from portable hand-held music players to set-top DVD players, support the playback of WMA files. Most PlaysForSure-certified online stores distribute content using this codec only. In 2005, Nokia announced its plans to support WMA playback in future Nokia handsets. In the same year, an update was made available for the PlayStation Portable (version 2.60) which allowed WMA files to be played on the device for the first time.
    Windows Media Audio Professional.

    Windows Media Audio Professional (WMA Pro) is a newer and more advanced lossy audio codec. It is based on a new compression algorithm which is not only superior to WMA in terms of quality, efficiency, and features, but also scales well at low bit rates. Its main competitors include AAC, HE-AAC, Vorbis, Dolby Digital, and DTS. It can support audio resolutions of up to 96 kHz and up to eight discrete channels (7.1 channel surround). WMA Pro also supports dynamic range compression, which reduces the volume difference between the loudest and quietest sounds in the audio track. According to Microsoft's Amir Majidimehr, WMA Pro can technically go beyond 7.1 surround sound and support "an unlimited number of channels."

    The codec's bit stream syntax was frozen at the first version, WMA 9 Pro Later versions of WMA Pro introduced low-bit rate encoding, low-delay audio, frequency interpolation mode, and an expanded range of sampling rate and bit-depth encoding options. A WMA 10 Pro file compressed with frequency interpolation mode comprises a WMA 9 Pro track encoded at half the original sampling rate, which is then restored using a new compression algorithm. In this situation, WMA 9 Pro players which have not been updated to the WMA 10 Pro codec can only decode the lower quality WMA 9 Pro stream. Starting with WMA 10 Pro, eight channel encoding starts at 128 kbit/s, and tracks can be encoded at the native audio CD resolution (44.1 kHz, 16-bit), previously the domain of WMA Standard.

    Despite a growing number of supported devices and its superiority over WMA, WMA Pro still has little hardware and software support. Some notable exceptions to this are the Microsoft Zune (limited to stereo), Xbox 360, Windows Mobile-powered devices with Windows Media Player 10 Mobile, and newer Toshiba Gigabeat and Motorola devices. In addition, WMA Pro is a requirement for the WMV HD certification program. On the software side, Verizon utilizes WMA 10 Pro for its V CAST Music Service, and Windows Media Player 11 has promoted the codec as an alternative to WMA for copying audio CD tracks. WMA Pro is supported in Silverlight as of version 2 (though only in stereo mode). In the absence of the appropriate audio hardware, WMA Pro can automatically downmix multichannel audio to stereo or mono, and 24-bit resolution to 16-bit during playback.

    A notable example of WMA Pro being used instead of WMA Standard is the NBC Olympics website which uses WMA 10 Pro in its low-bitrate mode at 48 kbit/s.
    Windows Media Audio Lossless

    Windows Media Audio Lossless (WMA Lossless) is a lossless audio codec that competes with ATRAC Advanced Lossless, Dolby TrueHD, DTS-HD Master Audio, Apple Lossless, Shorten, Monkey's Audio, FLAC, and WavPack (the last two have the advantage of being open source software and available for nearly any operating system). Designed for archival purposes, it compresses audio signals without loss of quality from the original using VBR. When decompressed, the audio signal is an exact replica of the original. The first version of the codec, WMA 9 Lossless, and its revisions support up to 96 kHz, 24-bit audio for up to 6 discrete channels (5.1 channel surround) with dynamic range compression control. The typical compression ratio for music varies between 1.7:1 and 3:1.

    Hardware support for the codec is available on the Cowon A3, Bang & Olufsen Serenata , Sony Walkman NWZ-A and NWZ-S series, Zune 4, 8, 80 30, Zune 120 (with firmware version 2.2 or later) and the new Zune HD, Xbox 360, Windows Mobile-powered devices with Windows Media Player 10 Mobile,Toshiba Gigabeat S and V models, Toshiba T-400, the Meizu M3, and Best Buy's Insignia NS-DV, Pilot, and Sport music players. Contrary to some claims, the Archos make of media devices do not support WMA Lossless, nor does the SONOS system. Like WMA Standard, WMA Lossless is being used by a few online stores to distribute music online. Similar to WMA Pro, the WMA Lossless decoder can perform downmixing when capable audio hardware is not present.
    Windows Media Audio Voice

    Windows Media Audio Voice (WMA Voice) is a lossy audio codec that competes with Speex (used in Microsoft's own Xbox Live online service), ACELP, and other codecs. Designed for low-bandwidth, voice playback applications, it employs low-pass and high-pass filtering of sound outside the human speech frequency range to achieve higher compression efficiency than WMA. It can automatically detect sections of an audio track containing both voice and music and use the standard WMA compression algorithm instead. WMA Voice supports up to 22.05 kHz for a single channel (mono) only. Encoding is limited to constant bit rate (CBR) and up to 20 kbit/s. The first and only version of the codec is WMA 9 Voice.

    Windows Mobile-powered devices with Windows Media Player 10 Mobile have native support for WMA 9 Voice playback. In addition, BBC World Service has employed WMA Voice for its Internet radio streaming service.
    Sound quality

    See codec listening test for a table of double-blind listening test results.

    Microsoft claims that audio encoded with WMA sounds better than MP3 at the same bit rate; Microsoft also claims that audio encoded with WMA at lower bit rates sound better than MP3 at higher bit rates. Double blind listening tests with other lossy audio codecs have shown varying results, from failure to support Microsoft's claims about its superior quality to supremacy over other codecs. One independent test conducted in May 2004 at 128 kbit/s showed that WMA was roughly equivalent to LAME MP3; inferior to AAC and Vorbis; and superior to ATRAC3 (software version).

    Some conclusions made by recent studies:

    * At 32 kbit/s, WMA Standard was noticeably better than LAME MP3, but not better than other modern codecs in a collective, independent test in July 2004.
    * At 48 kbit/s, WMA 10 Pro was ranked second after Nero HE-AAC and better than WMA 9.2 in an independent listening test organized and supported by Sebastian Mares and Hydrogenaudio Forums in December 2006. This test, however, used CBR for WMA 10 Pro and VBR for the other codecs.
    * At 64 kbit/s, WMA Pro outperformed Nero HE-AAC in a commissioned, independent listening test performed by the National Software Testing Labs in 2005. Out of 300 participants, "71% of all listeners indicated that WMA Pro was equal to or better than HE AAC."
    * At 80 kbit/s and 96 kbit/s, WMA had lower quality than HE-AAC, AAC-LC, and Vorbis; near-equivalent quality to MP3, and better quality than MPC in individual tests done in 2005.
    * At 128 kbit/s, there was a four-way tie between aoTuV Vorbis, LAME MP3, WMA 9 Pro and AAC in a large scale test in January 2006, with each codec sounding close to the uncompressed music file for most listeners.
    * At 768 kbit/s, WMA 9 Pro delivered full-spectrum response at half the bit rate required for DTS in a comparative test done by EDN in October 2003. The test sample was a 48 kHz, 5.1 channel surround audio track.

    Criticism of claimed quality

    Microsoft's claims of WMA sound quality have frequently drawn complaints. "Some audiophiles challenge Microsoft's claims regarding WMA's quality," according to a published article from EDN. Another article from MP3 Developments wrote that Microsoft's claim about CD-quality audio at 64 kbit/s with WMA was "very far from the truth." At the early stages of WMA's development, a representative from RealNetworks claimed that WMA was a "clear and futile effort by Microsoft to catch up with RealAudio 8"

    Microsoft has sometimes claimed that the sound quality of WMA at 64 kbit/s equals or exceeds that of MP3 at 128 kbit/s (both WMA and MP3 are considered near-transparent at 128 kbit/s by most listeners). In a 1999 study funded by Microsoft, NSTL found that listeners preferred WMA at 64 kbit/s to MP3 at 128 kbit/s (as encoded by MusicMatch Jukebox). However, a September 2003 public listening test conducted by Roberto Amorim found that listeners preferred 128 kbit/s MP3 to 64 kbit/s WMA audio with greater than 99% confidence. This conclusion applied equally to other codecs at the same bitrate, leading him to conclude that:
    “ No codec delivers the marketing plot of same quality as MP3 at half the bitrates.”

    It is important to note that both MP3 and WMA encoders have undergone active development and improvement for many years, so their relative quality may change over time.

    A July 2007 public listening test by Sebastian Mares found that 64 kbit/s HE-AAC audio (encoded by Nero Digital) was statistically tied with 64 kbit/s WMA Pro audio, in terms of listener preference.
    Players
    Screenshot of VLC Media Player which supports most of the Windows Media Audio compression formats.

    Apart from Windows Media Player, most of the WMA compression formats can be played using ALLPlayer, VLC media player,MPlayer, RealPlayer, Winamp, Zune Software (with certain limitations—DSP plugin support and DirectSound output is disabled using the default WMA plugin), and many other software media players. The Microsoft Zune media management software supports most WMA codecs, but uses a variation of Windows Media DRM which is used by PlaysForSure.

    The FFmpeg project has reverse-engineered and re-implemented the WMA codecs (except WMA Lossless and WMA Voice) to allow their use on POSIX-compliant operating systems such as Linux. The rockbox project further extended this codec to be suitable for embedded cores, allowing playback on portable MP3 players and cell phones running open source software. RealNetworks has announced plans to support playback of DRM-unprotected WMA files in RealPlayer for Linux. On the Macintosh platform, Microsoft released a PowerPC version of Windows Media Player for Mac OS X in 2003, but further development of the software has ceased. Microsoft currently endorses the third-party Flip4Mac WMA, a QuickTime component which allows Macintosh users to play WMA files in any player that uses the QuickTime framework. Flip4Mac however does not currently support the Windows Media Audio Voice codec.
    Encoders

    Software that can export audio in WMA format include Windows Media Player, Windows Movie Maker, Microsoft Expression Encoder, GOM Player, RealPlayer, Adobe Premiere Pro, Adobe Audition, and Adobe Soundbooth. Microsoft Office OneNote supports encoding in all WMA codecs, and Windows Media Encoder supports all available bit rate and resolution options as well.

    Digital rights management

    While none of the WMA codecs themselves contain any DRM facilities, the ASF container format, in which a WMA track may be encapsulated, can. Windows Media DRM, which can be used in conjunction with WMA, supports time-limited music subscription services such as those offered by unlimited download services, including MTV's URGE, Napster, Rhapsody, Yahoo! Music Unlimited and Virgin Digital. Windows Media DRM, a component of PlaysForSure and Windows Media Connect, is supported on many modern portable audio devices and streaming media clients such as Roku, SoundBridge, Xbox 360 and Wii. Players that support the WMA format but not Windows Media DRM list protected titles as unplayable.

    Above Article from the Wikipedia



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    Advanced Audio Coding (AAC) format

    Post by maxim9691 on Mon Jan 25, 2010 1:09 pm

    Advanced Audio Coding (AAC) format

    Advanced Audio Coding (AAC) is a lossy digital audio compression format. Because of its exceptional performance and quality, AAC is at the core of the MPEG-4, 3GPP and 3GPP2 specifications and is the audio codec of choice for Internet, wireless and digital broadcast arenas.

    AAC was designed as an improved-performance codec relative to MP3 (which was specified in MPEG-1 and MPEG-2) by the ISO/IEC in 11172-3 and 13818-3.

    AAC provides audio encoding that compresses much more efficiently than older formats, such as MP3 or OGG, yet delivers quality rivaling that of uncompressed CD audio.
    Features of AAC format

    AAC was promoted as the successor to MP3 for audio coding at medium to high bitrates. When compared side-by-side, AAC proves itself worthy of replacing MP3 as the new Internet audio standard the advantages over MP3 are:

    * Improved compression provides higher-quality results with smaller file sizes.
    * Support for multichannel audio, providing up to 48 full frequency channels.
    * Higher resolution audio, yielding sampling rates up to 96 kHz.
    * Improved decoding efficiency, requiring less processing power for decode.

    History of AAC

    Advanced Audio Coding was developed with the cooperation and contributions of companies mainly including Dolby, Fraunhofer (FhG), AT&T, Sony and Nokia, and was officially declared an international standard by the Moving Pictures Experts Group in April of 1997.

    AAC was written into specification as Part 7 of the MPEG-2 standard, and again into Part 3 of the MPEG-4 standard. As such, AAC can be referred to as MPEG-2 Part 7 and MPEG-4 Part 3 depending on its implementation, but is most often referred to as MPEG-4 AAC, or AAC for short.

    However, MP4 usually refers to the format described in MPEG-4 Part 14, which is a container format for carriage of video and audio data.



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    Ogg Vorbis

    Post by maxim9691 on Mon Jan 25, 2010 1:17 pm

    Ogg Vorbis. A fairly popular open source compression format reputed as having very good sound quality. Somewhat limited player support, however.

    INTRODUCTION
    Ogg Vorbis will likely be considered one of the most exciting developments in music history. As a creator of music, nothing is more important than getting your music heard. But, how can a creator of music be sure that the music he produces today isn't restricted in the future?

    With the advent of MP3, I thought this problem was solved. It was not until this year that I came to a full understanding of what freedom really means. As it turns out, several individuals and companies plan to restrict the use of MP3's. Starting this year some of the patent holders to particular MP3 technologies plan on enforcing their property rights. As a creator of music, I will be restricted in my efforts to "get the music heard."

    Ogg Vorbis to the rescue! Ogg Vorbis is truly open source and royalty free. A musician can create Ogg files and never have to worry about the music being restricted or money being extorted!

    "Oh, all the kids are switching over to Ogg. The kids are smart. They know it is a better file format than MP3 -- it is faster to download and requires less storage space."
    -- Dr. Sidd Mukherjee, Ohio State

    INTERVIEW
    Chris "Monty" Montgomery is accused of bearing the brain child known as Ogg Vorbis. He has agreed to this rock n' roll style interview about digital music:

    Q: Monty, how long have you been working on Ogg Vorbis?

    The beginnings of Ogg, before it was named Ogg, were in 1993. Vorbis and the Ogg code of today started in fall of 1998 shortly after Fraunhofer decided to 'take back' mp3 and threatened to sue all the free mp3 projects out of existence.

    Q: What made you decide to work on a music related project... don't you know that there is no money for the creators in the music business?

    If I cared about money, I'd be retired already and terminally bored.

    I'm a musician, and used to be a serious one (serious, but not very good). I wanted to mix and master recordings digitally on my new, unbelievably powerful 486DX50 running Linux, but I only had about 800M of disk, the biggest disk I could afford.

    Ogg started because I needed it, it was interesting, and I wasn't going to find it anywhere else.

    Mp3 existed then, but the only encoders worth using were impossibly expensive for a student. The dist10 encoder was useless. Blade wasn't much better. LAME didn't exist. It was Fraunhofer mp3enc or nothing, and I didn't have the money, so it was nothing.

    Nothing turned into Ogg, and Ogg turned out to be a lot more interesting than the music I was working on.

    Q: How massive of a project has it been? Can you tell me more about the people that have helped you?

    Not a massive project, just a long one. People who have been involved have been friends who also found it interesting and other on the net who also needed Ogg, but needed some part that didn't exist yet. Good hackers don't complain when they need something that doesn't exist, they write what they need.

    Q: I'd hate to see Ogg Vorbis go away. How can it be financially viable in the long run? How will it support itself?

    Ogg Vorbis will never go away. The code is there, the format is there and nothing will ever undo that. I expect you mean that you want it to keep getting better....

    I have few worries about seeing Ogg improve. Right now, there are enough individuals who both want to see it improve and have, themselves, the skill to improve it. At some point, the consumer electronic, software and music industries will also see that the continuance of Ogg is in their interests, and they will contribute to Ogg as well, even if it's not necessarily directly to the core Free project.

    Until that critical mass happens, we'll probably do just fine on the contributions from people who are mostly interested in the tax write-off :-)

    Q: Ogg Vorbis uses a technology called VBR (variable bit rate).

    It's not so much a technology as a description. It's like calling the color red a technology.

    This is great for allowing listeners a smoother listening experience over the internet. But, another important role of VBR is conservation, correct? I mean will overall server space and bandwidth be reduced making the Internet a better place for everyone?

    Size and quality are always related in a lossy perceptual codec like mp3, WMA or Ogg. When a compression, like Ogg, improves on the current state of the art, it fits more information into a physical number of bits. So you can put it either way: "Ogg files can be smaller and sound just as good", or "Ogg files sound better for the same size."

    What VBR does is prevent wasting bits on stupidly encoding unnecessary information just because we *must* fill the space *now*. We can use the saved space later for harder parts (increasing quality for the same size), or we can hoard the savings (decreasing size for the same quality).

    Q: The name Ogg Vorbis -- is there any chance that you will place restrictions on using the name in the future? Though I understand the technology is to be patent free, does this also mean that I needn't ever worry about having to rename all my .ogg files in the future?

    We will not restrict the name, but we will smack anyone who creates something that's not compatible and calls it 'Ogg Vorbis'. Anyone who builds a workalike is welcome to use the name (and the logo).

    Q: We've been experimenting with the fidelity of .ogg files. Our latest experiment involves a "live album." It was recorded as a 40 minute long, 48kHz stereo .wav file. Can I achieve the highest "audible" fidelity by encoding using the default settings on the Ogg encoder? Or, should try a setting it higher than 128 kbps?

    As of beta 4, I personally encode at 160kbps. Because 1.0 has channel coupling, the encoder gets lots more 'free' bits to work with, so the unreleased test encoder I have right now I use at 96kbps and 128kbps for 'hi fi'.

    Q: As a follow up to that question, will the Ogg Vorbis format only use as many kbps as is necessary to get the highest fidelity? If so, shouldn't I encode all files at the highest kbps setting available? (which is 350 kbps on the encoder that I am using.)

    beta 4 is actually using 'quality modes' that are pure VBR and just happen to average particular bitrates. In 1.0, you'll be able to select real bitrates or these pure VBR quality modes.

    The various quality modes progressively 'back off' the psychoacoustic model so that, for higher quality, the model is increasingly conservative. The reason is that no model, including the very good one in Vorbis, is perfect, and mistakes might be audible.

    By using a higher quality mode, you reduce the likelihood of the model making an audible mistake. The tradeoff is that it uses more bits.

    Q: Part three of this question is -- will the file size always be directly proportional to the length in time of a music file. (i.e. If a 15 minute piece of music is approximately a 12.5M Ogg file (at 128kbps), will a 30 minute piece be about 25M?)

    For the same bitrate, yes. Bitrate is not a measurement of quality, it's a measure of size even if our own current encoder fudges the distinction. 1.0 will not fudge.

    Q: Our goal is to have a music video for every song. However, we have not been able to find a comparable file format to Ogg Vorbis for video. (The next best thing that we can find is MPEG2.)

    Yeah, we know.

    Do you know if such a beast exists? (I've heard rumors that your group is working on a project called "Tarkin.")

    We are indeed working on a project called Tarkin. The project is only in a research stage, and there is no released code (and no release currently planned). Aside from being technically very different from audio, the video patent minefield is also currently harder to navigate.

    Q: And, what about the name? How do you say it? Like egg nog... egg ogg?

    Yup, that works.

    Monty

    Interview by Membrane.com



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    ATRAC. Sony's proprietary compression format.

    Post by maxim9691 on Mon Jan 25, 2010 1:20 pm

    ATRAC. Sony's proprietary compression format. If you are going to buy a Sony player, such as their Network Walkman (NW-HD1), you can only use this format. More on this in the player guide section - remember beta video tapes, anyone?

    Sony has made an announcement that it is ending support for its proprietary ATRAC format.

    Sony issued a press release, which, on the surface, seemed to herald an excitement-filled event--the release of two new multi-featured Walkman players, which would include video support for the first time.

    While this was an announcement of some significance and anticipation, the carefully worded fine print of the release sounded a far different tune for Sony.

    Near the end of the release, Sony stated that:

    All of the new players are compatible with security-enhanced Windows Media Audio and support most subscription music services

    It goes on to casually mention:

    As a result, Sony will be phasing out the CONNECT™ Music Services based on Sony's ATRAC audio format in North America and Europe. Specific timing will vary by region depending on market demand, but will not be before March 2008. The CONNECT e-book service for the Reader will not be affected.

    While Sony's CONNECT store has by no means been a commercial hit, it is surprising to hear that Sony will not be providing ATRAC support as its Walkman players have featured support for the various ATRAC codecs, since the format's release in the 90s.

    Matt Moore of the Associated Press further elaborated on ATRAC's fate in an article following this press release:

    Sony spokeswoman Linda Barger said the new Walkman players will no longer directly support ATRAC.

    We are offering conversion software to convert ripped non-secure ATRAC files to MP3,' she said in an e-mail.'

    From the now defunct Betamax, to today's Blu-Ray Disc format, Sony has always invested heavily in developing and commercially promoting its own proprietary formats.

    ATRAC was no exception. ATRAC, which stands for Adaptive Transform Acoustic Coding, is a set of audio compression algorithms, akin to the MP3 compression format. It was first commercially released in 1992, with Sony's Walkman players, which used the format to compress audio for the now defunct Mini-Disc format. As well as providing compression, ATRAC engineer's designed it with the intention of encoding audio at high speeds with minimal power consumption. Since its inception, ATRAC has seen four iterations developed by Sony -- ATRAC1, ATRAC3, ATRAC3plus, and ATRAC Advanced Lossless.

    Sharp and Panasonic have provided third party support in the past, for the format, creating their own implementations of the ATRAC codec, for use in their Mini-Disc players.

    With Sony's announcement, it appears that this format is finally approaching extinction.

    Curiously, Sony's webpage for the ATRAC format makes no reference to these developments.

    Sony's Connect store's impending closure may be slightly less surprising, as the store is still relatively new, having just been created in 2005, and has been the victim of many problems. The store was initially intended to provide a service similar to iTunes.

    CNET's John Borland released an excellent story on the history of CONNECT's problems, titled "How Sony failed to Connect, again". It chronicles how the service has been plagued by problems since its creation, stemming from internal disputes among the development team and bug-prone software releases, which yielded a large amount of negative customer-feedback.

    Aside from the issues of attempting to deal with the difficulties of promoting a proprietary format and dealing with the Connect store's problems, Sony's biggest problem in the portable music industry, has simply been weak sales of its Walkman® player line.

    Bloomberg, who compiled a list of the top electronics retailers in various sectors by market share, for the month of March 2007, indicated that Sony was not among the top five sellers in the MP3 player market. The top five sellers, respectively, were Apple, Sansdisk, Creative Labs, Microsoft and Samsung. Similarly, The NPD Group released sales figures for flash-memory MP3 player sales, which revealed that Sony had no players that were among the ten highest-selling models for the year through June.

    With the demise of Connect and ATRAC impending, Sony, once the clear leader in portable audio, faces an uncertain future in this sector, as it attempts a tough uphill battle to regain significant market share, in one of electronics industry's most competitive markets.

    Above Article by [You must be registered and logged in to see this link.]



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    Apple Lossless

    Post by maxim9691 on Mon Jan 25, 2010 1:23 pm

    And now the Lossless

    Apple Lossless. Apple's proprietary lossless format is getting good reviews, but again, you are pretty much restricted to Apple products.

    Apple Lossless (also known as Apple Lossless Encoder, ALE, or Apple Lossless Audio Codec, ALAC) is an audio codec developed by Apple Inc. for lossless data compression of digital music.

    Apple Lossless data is stored within an MP4 container with the filename extension .m4a. It is not a variant of AAC, but uses linear prediction similar to other lossless codecs such as FLAC and Shorten. [1] All current iPod players can play Apple Lossless-encoded files. It does not utilize any digital rights management (DRM) scheme, but by the nature of the container, it is thought that DRM could be applied to ALAC much the same way it can with other files in QuickTime containers.

    Apple claims that audio files compressed with its lossless codec will use up "about half the storage space" that the uncompressed data would require. Testers using a selection of music have found that compressed files are about 40% to 60% the size of the originals depending on the kind of music, similar to other lossless formats. Furthermore, the speed at which it can be decoded makes it useful for a limited-power device such as the iPod. [2]

    The Apple Lossless Encoder was introduced as a component of QuickTime 6.5.1 on April 28, 2004 and thus as a feature of iTunes 4.5 and above. The codec is also used in the AirPort Express's AirTunes implementation.

    David Hammerton and Cody Brocious have analyzed and decoded this codec without any documents on the format. On March 5, 2005, Hammerton published a simple open source decoder in the programming language C on the basis of the reverse engineering work.

    The open source library libavcodec now incorporates both a decoder and an encoder for Apple Lossless format which means that media players based on that library, including VLC media player and MPlayer, are able to play Apple Lossless files.

    Above Article by Wapedia



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    FLAC (Free Lossless Audio Codec)

    Post by maxim9691 on Mon Jan 25, 2010 1:26 pm

    FLAC stands for Free Lossless Audio Codec, an audio format similar to MP3, but lossless, meaning that audio is compressed in FLAC without any loss in quality. This is similar to how Zip works, except with FLAC you will get much better compression because it is designed specifically for audio.
    It differs from MP3, WMA, MP4 and other formats with lossy compression because it does not cause quality loss. However, the compression ratio is lower.

    FLAC is freely available and supported on most operating systems, including Windows, UNIX (Linux, *BSD, Solaris, OS X, IRIX), BeOS, OS/2, and Amiga. There are build systems for autotools, MSVC, Watcom C, and Project Builder.

    The FLAC project consists of:

    * the stream format
    * reference encoders and decoders in library form
    * flac, a command-line program to encode and decode FLAC files
    * metaflac, a command-line metadata editor for FLAC files
    * input plugins for various music players

    When we say that FLAC is free it means more than just that it is available at no cost. It means that the specification of the format is fully open to the public to be used for any purpose (the FLAC project reserves the right to set the FLAC specification and certify compliance), and that neither the FLAC format nor any of the implemented encoding/decoding methods are covered by any known patent. It also means that all the source code is available under open-source licenses. It is the first truly open and free lossless audio format.

    Some claim FLAC is the most widely used lossless compression format on UNIX systems (though it seems more likely that shn retains that honor on all OS platforms). FLAC files also can be placed inside an Ogg container using libOggFLAC and libOggFLAC++.
    Features

    * Lossless: The encoding of audio (PCM) data incurs no loss of information, and the decoded audio is bit-for-bit identical to what went into the encoder. Each frame contains a 16-bit CRC of the frame data for detecting transmission errors. The integrity of the audio data is further insured by storing an MD5 signature of the original unencoded audio data in the file header, which can be compared against later during decoding or testing.
    * Fast: FLAC is asymmetric in favor of decode speed. Decoding requires only integer arithmetic, and is much less compute-intensive than for most perceptual codecs. Real-time decode performance is easily achievable on even modest hardware.
    * Hardware support: Because of FLAC's free reference implementation and low decoding complexity, FLAC is currently the only lossless codec that has any kind of hardware support.
    * Streamable: Each FLAC frame contains enough data to decode that frame. FLAC does not even rely on previous or following frames. FLAC uses sync codes and CRCs (similar to MPEG and other formats), which, along with framing, allow decoders to pick up in the middle of a stream with a minimum of delay.
    * Seekable: FLAC supports fast sample-accurate seeking. Not only is this useful for playback, it makes FLAC files suitable for use in editing applications.
    * Flexible metadata: New metadata blocks can be defined and implemented in future versions of FLAC without breaking older streams or decoders. Currently there are metadata types for tags, cue sheets, and seek tables. Applications can write their own APPLICATION metadata once they register an ID.
    * Suitable for archiving: FLAC is an open format, and there is no generation loss if you need to convert your data to another format in the future. In addition to the frame CRCs and MD5 signature, flac has a verify option that decodes the encoded stream in parallel with the encoding process and compares the result to the original, aborting with an error if there is a mismatch.
    * Convenient CD archiving: FLAC has a cue sheet metadata block for storing a CD table of contents and all track and index points. For instance, you can rip a CD to a single file, then import the CD's extracted cue sheet while encoding to yield a single file representation of the entire CD. If your original CD is damaged, the cue sheet can be exported later in order to burn an exact copy.
    * Error resistant: Because of FLAC's framing, stream errors limit the damage to the frame in which the error occurred, typically a small fraction of a second worth of data. Contrast this with some other lossless codecs, in which a single error destroys the remainder of the stream.

    Pros

    * Portable to many systems
    * Source open and freely licenced
    * Hardware support (PhatBox, Kenwood MusicKeg, Rio Karma, etc. See below)
    * Streaming support
    * Extremely fast decoding
    * Supports multichannel and high resolution streams
    * Supports Replay Gain
    * Supports cue-sheet (with some limitations)
    * Gaining wide use as successor to Shorten

    Cons

    * Compresses less efficiently than other popular modern compressors (Monkey's Audio, OptimFROG)
    * Higher compression modes slow, for little gain over the default setting.



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    Monkey’s Audio (APE)

    Post by maxim9691 on Mon Jan 25, 2010 1:29 pm

    Monkey’s Audio is a file format for audio data compression. Being a lossless format, Monkey’s Audio does not discard data during the process of encoding, unlike lossy compression methods such as AAC, MP3, Vorbis and Musepack.

    Data file compression is employed in order to reduce bandwidth, file transfer time, and/or storage requirements. A digital recording (such as a CD) encoded to the Monkey’s Audio format can be decompressed into an identical copy of the original audio data. As with the FLAC and Apple Lossless format, files encoded to Monkey’s Audio are typically reduced to about half of the original size, with data transfer rates and bandwidth requirements being reduced accordingly.

    Monkey’s Audio's drawbacks are that it is proprietary software, and has limited support on software platforms other than Windows. There are alternatives such as FLAC and WavPack that may offer more options for some users.

    Monkey’s Audio files use the filename extension .ape for audio, and .apl for track metadata.

    Monkey’s Audio generally achieves compression rates which are slightly better than FLAC and significantly better than Shorten (an older format no longer in development). Both encoding and decoding are generally somewhat slower than with FLAC or Shorten, and the decoder is problematic to implement on portable digital audio players. It suffers from relatively slow seeking, depending on the compression level chosen.

    Monkey’s Audio is freeware. It has a peculiarly ambiguous non-free software license, thus most Linux distributions and other operating systems that rely on free software alone do not include i and the software environment around Monkey’s Audio is less varied than that of other, more freely-licensed lossless compressors such as FLAC. FLAC comes pre-installed with most Linux distributions and is typically preferred by Linux users. The Shorten format is still in common use on sites such as etree, having been the de facto standard of live taping enthusiasts for years, but the advantage of FLAC as a broad-based, mature format being continuously refined by active development is prompting some to make the switch.

    Since Monkey’s Audio is a lossless compression method, it is not readily comparable with lossy compression schemes such as AAC, MP3 and Vorbis. The two types of formats have different aims: lossless compression seeks to exactly preserve the original file in as small a space as possible, and lossy codecs are designed to retain sufficient audio quality to satisfy less demanding reproduction standards, through discarding varying amounts of sound data in order to shrink the file to fit the allotted file space. This is done by adjusting the sampling/bit rate, and in turn, the compression ratio. Therefore, audio files compressed using lossless audio formats are significantly larger than ones encoded in lossy formats. A typical Monkey’s Audio file will be approximately 3-5 times larger than a 192kb/second bitrate MP3.
    Supported platforms

    Officially, Monkey’s Audio is available only for the Microsoft Windows platform, though discussion on the Monkey’s Audio website has hinted at future support for Linux and Mac OS. A developer using the moniker SuperMMX released an unofficial port in early 2005, which also includes plugins to allow playback using Beep Media Player and the XMMS audio player. This port was originally developed for Linux but, since version 3.99 update 4 build 4, it has included support for Mac OS X as well as Linux on the PowerPC and SPARC architectures.

    While the license text claims to permit using the official Monkey's Audio codec in GPL projects, several Linux distribution maintainers have found the license to be contradictory. It does not permit redistribution or modification, and thus is not considered open source or free software.

    Monkey's Audio files can be encoded and decoded on any platform which has a J2SE implementation, by the means of the unofficial JMAC library, which is Free software licensed under the GNU LGPL.

    On hardware platforms, the open source jukebox firmware project Rockbox supports playback of Monkey's Audio-encoded files on most of its supported targets, but many lack sufficient processing power to play the files back in real time on any but the lowest compression settings. In addition, there are a few MP3 players that natively support Monkey's Audio, including the Cowon D2 and the Iriver Spinn. As of version 4.02 (January 19, 2009) a directshow filter is distributed with the installer, allowing for compatibility with most mp3 players

    APEv2

    The creator of the Musepack format Frank Klemm extended the original format to add a header, allowing APE tags to be at the beginning of files, and also allowed metadata values to be Unicode rather than simply ASCII. Because of its simplicity and flexibility, the WavPack and OptimFROG formats also adopted it as their primary tag format. Version 3.99 of Monkey's Audio also switched from using APEv1 to APEv2.

    foobar2000 and Winamp can also tag MP3 files with APEv2 tags instead of ID3 tags. Some music library managers and mass taggers (MusicBee, Mp3tag) also support this feature. However, because ID3 was designed with the MP3 format in mind and APEv2 was not, there are some complications. For example, the string APETAGEX signals the start of an APEv2 tag, but the string TAG signals the start of an ID3v1 tag. If the TAG in APETAGEX ends up where an ID3v1 tag is expected, it may be read incorrectly. ID3 also has an “unsynchronization scheme” to ensure players do not try to play the tag data as audio; APEv2 has no such scheme, and APEv2 tags may result in errors or static at the end of files.
    Features

    APE tags are closer to Vorbis comments than ID3 tags. Like Vorbis comments, they are unstructured key/value pairs. However, unlike Vorbis comments, they do not allow for inter-key ordering. This is because they store a list of values with each key, rather than a key per value.

    APE values can also be flagged as text, binary, or external types. Because of this, tag editing software can avoid presenting blocks of unreadable data to users. Binary data cannot be easily stored in Vorbis comments for this reason (though this is by design).

    APEv2 supports Unicode using UTF-8 for key values. For key names, an ASCII subset (control characters from 0x00 to 0x1f are not allowed) must be used.

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    iPods and Young People Have Utterly Destroyed Music

    Post by maxim9691 on Thu Mar 04, 2010 10:41 am

    You know how most people are perfectly happy with Apple standard-issue earbuds, white plastic molded around a crappy audio experience? A Stanford professor's informal annual study shows that youngins like the "sizzle sounds" of MP3s.

    Each year, Stanford Professor of Music Jonathan Berger does an informal test of his students by playing a bunch of different music in a bunch of different formats. Over email, here's how he told me performs the informal study:

    Students were asked to judge the quality of a variety of compression methods randomly mixed with uncompressed 44.1 KHz audio. The music examples included both orchestral, jazz and rock music. When I first did this I was expecting to hear preferences for uncompressed audio and expecting to see MP3 (at 128, 160 and 192 bit rates) well below other methods (including a proprietary wavelet-based approach and AAC). To my surprise, in the rock examples the MP3 at 128 was preferred. I repeated the experiment over 6 years and found the preference for MP3 - particularly in music with high energy (cymbal crashes, brass hits, etc) rising over time.

    In other words, younger people haven't just grown more tolerant of thin, soulless MP3 renditions of their favorite music, they actually like them. Shitty MP3s, even. O'Relly Radar quotes Professor Berger as saying that it's the "sizzle sounds" that people are loving because it's what they're comfortable with. So, yes Virginia, iPods really have killed music. People aren't just ignorant of high quality audio, they actually hate it. Gee, thanks for contributing to the downfall of civilization, Apple. Music is dead, everyone, carry on.

    Send an email to matt buchanan, the author of this post, at [You must be registered and logged in to see this link.].



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    Re: Digital Audio Format Guide "which is best for you?"

    Post by Ymaginatif on Thu Mar 04, 2010 10:46 am

    Then again ... young people used to listen to half-chewed-up cassette tapes. On stereos which had only one speaker attached. Or on cheap portable radios with bad reception. etc. etc.
    I think the lack of quality-control among young people is much older than the MP3!

    In fact, it's up to us 30plussers to keep the quality level of the audio-experience up!!!



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    Re: Digital Audio Format Guide "which is best for you?"

    Post by ElCyC on Thu Oct 14, 2010 3:03 pm

    The article on the professor doesn't surprise me but then again Ymaginatif's comment is true too ;-)

    Personally I think that also in terms of recent music it is catered for the mp3 generation and the boom boxes f.ex. listen to Black Eyed Peas on a decent ce-player vs a boombox.
    Also compression or loudness (I don't know how to call it exactly but today it's hard to find tracks where you really have soft passages and then loud, it all seems cranked up to the max) is the trend today.

    But then again, it's all about enjoying the music how and when you like

    cheers

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    Re: Digital Audio Format Guide "which is best for you?"

    Post by SeeNote on Sat Feb 25, 2012 9:04 pm

    I personally prefer the mp3 format due to it's tote-ability, as it's easily transferable to most mobile listening devices. I'd love have the lot of my music on AAC for quality, but it's pretty much restricted to iTunes, right?
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    Re: Digital Audio Format Guide "which is best for you?"

    Post by ElCyC on Sun Oct 07, 2012 9:15 am

    Will be interesting to see what come from Neil Young's battle for HQ audio "Pono".
    By the way anyone have any experience with so-called audiophile DAP's like Ibasso, Colorfly, ... ?

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    Re: Digital Audio Format Guide "which is best for you?"

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