MP3 Encoding

Mp3 is crap, WMA is the best!!

 

The bitrate is quality, you can only do maybe 320Kbps with Any of your decoders. I was able to go up to 980kbps or somewhere around there with windows media player. So if you can show me a file with 980kbps that you made with an encoder or decoder then i might beleive you lol. The bitrate is how you measure quality, Kbps You should't need any formulas to see lol. Even a wma at normal bitrate is far superior to mp3. Check the specs-bitrate. Kbps

 

Kbps

http://searchnetworking.techtarget.com/sDefinition/0,,sid7_gci212436,00.html


Bandwidth

http://searchnetworking.techtarget.com/sDefinition/0,,sid7_gci211634,00.html

 

I got your Allison Krauss data Ranger. God i love her voice!

Will try an a/b comparison later this evening, between wma, mp3, and the original source. I will only be able to judge it using my right ear right now, so my opinon really wont be worth a hoot!

Hopefully in a few weeks my ear drum will heal up and i can do some REAL critical listening

 

Hey, guys i downloaded Tom Sawyer as an MP3 at 128kbps and converted it to a WMA with windows media player. Ended up with a file of 26.3MB at 841Kbps. This is the best you will get if you want high quality. It sounds like you have a nice collection Ranger, I would back it up to a data dvd. If you burn them as a data dvd as wma at say 15mb each song, you would be able to fit 286 songs that are converted to WMA and are roughly 580Kbps right off your cds using windows media player. It converts very fast too, so you could do your whole collection, rip it into your media player library, if your hard drive is big enough, Then start burning data dvds with 286 high quality wma files on each dvd. Now you have a master backup and can do anything later on.

 

interesting electro. how do you extract more info from a lower bit rate source and get higher bit rates from it? The way i understand it is if you are given say, 1000 bits, you cannot get anymore out of it, like 3000 bits as an example. I am not up to date on this type of recording media, so an explanation would really help here!! Thanks!

 

The quality of a song is mostly still there but hiding....no matter how much you compress-convert it. They started doing this because of the internet being so slow when everyone had dial up modems there was a need to compress these huge wave files so people could download them. Now with cable, dsl and wireless, size of a file is getting to be less of a problem. Its all in the encoder, decoder. Windows Media Player has the best encoder decoder i have seen ever. It is the only one that can bring back a compressed song almost to it's original state. Winzip is very similar to an mp3 or wma as all it is doing is compressing information. Winrar is another way of compressing data. windows xp uses NTFS file system to compress data.

http://www.ntfs.com/

here's a pic of tom sawyer through the wave spectrum analyiser ! lol, it may be hard to see but if i stretched out the signal there is huge differences, much more fuller on the wma file. The top pic is of mp3 and the bottom is wma

 

This sounds very similiar to DBX dynamic range compression, albeit, its in the analog domain......versus the digital domain of this new stuff. Interesting. So the very basic info is there in all of the recordings, just that some sources extract more than others?

 

Data Compression

http://en.wikipedia.org/wiki/Data_compression

 

Yes, its all there, dbpoweramp and audiograbber are like aftermarket parts-software, Mikrosoft is king, OE lol thats why windows media player rules, i think they know a little about compression!! They have been there since the begining.

 

it started out as 10 to 1.0 now it's 1.7 to 1.0

http://www.caraudiotalk.com/audio-forum/attachment.php?attachmentid=206&d=1168742735

http://www.caraudiotalk.com/audio-forum/attachment.php?attachmentid=208&d=1168742735

 

Hey Ranger, sorry for thrashing out at you lol, in the message that i deleted, I was very irritated that day.
You have a lot of cds and it would take weeks maybe months? doing it song by song with your decoders. Windows Media Player will rip all your cds into your computer in no time.
All you have to do is change your burn options in the settings to the quality you want once, then click apply and save. Then any time you put an audio cd in your computer it will open up media player and you would just have to click rip to library. It takes maybe 10 minutes to copy one cd into your library.
Now after you do about 10 or so cds you can start burning the wma files on dvds as data dvds from your library. One dvd is 4.3GB which is 4,300 MB
So say at highest quality, lets say they are 25mb each song @ 980kbs well you can do the math! lol This will save you a HUGE amount of time and you will have a better result.

 

OK, where do I get those nice little recording/boards/mixer/whatever the hell else they do things!!!!

 

LOL, that has nothing to do with this thread really, was just showing what i use when i am doing recording live music or midi or digital, keyboards, whatever source. Here is the website:

http://www.propellerheads.se/products/reason/

 

Ranger...come back lol

MP3 is an audio-specific compression format. It provides a representation of pulse-code modulation-encoded audio in much less space than straightforward methods, by using psychoacoustic models to discard components less audible to human hearing, and recording the remaining information in an efficient manner. Similar principles are used by JPEG, a lossy image compression format.

The MP3 format uses a hybrid transformation to transform a time domain signal into a frequency domain signal:

32-band polyphase quadrature filter.
36 or 12 tap MDCT; size can be selected independently for sub-bands 0...1 and 2...31.
Aliasing reduction postprocessing.
MP3 audio can be compressed with several different bit rates, providing a range of tradeoffs between data size and sound quality.

The MPEG specifications support Advanced audio coding (AAC) from MPEG-4 as MP3's successor, although other new audio formats have also achieved similar usage levels. However, MP3's extreme popularity makes it secure in its dominant position for the near future, with support from a huge range of software and hardware, including portable MP3 players and even some DVD and CD players. The large MP3 collections that many individuals have amassed will also ensure its longevity, in the same way as with any physical medium.

MPEG-1 Audio Layer 2 encoding began as the Digital Audio Broadcast (DAB) project managed by Egon Meier-Engelen of the Deutsche Forschungs- und Versuchsanstalt für Luft- und Raumfahrt (later on called Deutsches Zentrum für Luft- und Raumfahrt, German Aerospace Center) in Germany. This project was financed by the European Union as a part of the EUREKA research program where it was commonly known as EU-147. EU-147 ran from 1987 to 1994.

In 1991, there were two proposals available: Musicam (known as Layer 2), and ASPEC (Adaptive Spectral Perceptual Entropy Coding). The Musicam technique, as proposed by Philips (The Netherlands), CCETT (France) and Institut für Rundfunktechnik (Germany) was chosen due to its simplicity and error robustness, as well as its low computational power associated to the encoding of high quality compressed audio. The Musicam format, based on sub-band encoding, was a key to settle the basis of the MPEG Audio compression format (sampling rates, structure of frames, headers, number of samples per frame). Its technology and ideas were fully incorporated into the definition of ISO MPEG Audio Layer I and Layer II and further on of the Layer III (MP3) format. Under the chairmanship of Professor Mussmann (University of Hannover) the editing of the standard was made under the responsibilities of Leon van de Kerkhof (Layer I) and Gerhard Stoll (Layer II).

A working group consisting of Leon Van de Kerkhof (The Netherlands), Gerhard Stoll (Germany), Leonardo Chiariglione (Italy), Yves-François Dehery (France), Karlheinz Brandenburg (Germany) took ideas from Musicam and ASPEC, added some of their own ideas and created MP3, which was designed to achieve the same quality at 128 kbit/s as MP2 at 192 kbit/s.

All algorithms were approved in 1991, finalized in 1992 as part of MPEG-1, the first standard suite by MPEG, which resulted in the international standard ISO/IEC 11172-3, published in 1993. Further work on MPEG audio was finalized in 1994 as part of the second suite of MPEG standards, MPEG-2, more formally known as international standard ISO/IEC 13818-3, originally published in 1995.

Compression efficiency of encoders is typically defined by the bit rate because compression rate depends on the bit depth and sampling rate of the input signal. Nevertheless, there are often published compression rates that use the CD parameters as references (44.1 kHz, 2 channels at 16 bits per channel or 2x16 bit). Sometimes the Digital Audio Tape (DAT) SP parameters are used (48 kHz, 2x16 bit). Compression ratios with this reference are higher, which demonstrates the problem of the term compression ratio for lossy encoders.

In October 1993, MP2 (MPEG-1 Audio Layer 2) files appeared on the Internet and were often played back using the Xing MPEG Audio Player, and later in a program for Unix by Tobias Bading called MAPlay, which was initially released on February 22, 1994 (MAPlay was also ported to Microsoft Windows).

Initially the only encoder available for MP2 production was the Xing Encoder, accompanied by the program CDDA2WAV, a CD processor that transforms CD audio tracks to Waveform Audio Files.

The Internet Underground Music Archive (IUMA) is generally recognized as the start of the on-line music revolution. IUMA was the Internet's first high-fidelity music web site, hosting thousands of authorized MP2 recordings before MP3 or the web was popularized.

In the first half of 1995 through the late 1990s, MP3 files began flourishing on the Internet. MP3 popularity was mostly due to, and interchangeable with, the successes of companies and software packages like Nullsoft's Winamp (released in 1997), mpg123, and Napster (released in 1999). Those programs made it very easy for the average user to playback, create, share, and collect MP3s.

Controversies regarding peer-to-peer file sharing of MP3 files have spread widely in recent years — largely because high compression enables sharing of files that would otherwise be too large and cumbersome to easily share. Some major record companies reacted by filing a lawsuit against Napster, due to the vastly increased spread of MP3s through the Internet, to protect their copyrights (see also intellectual property).

 

Commercial online music distribution services (like the iTunes Store) usually prefer other/proprietary music file formats that support Digital Rights Management (DRM) to control and restrict the use of digital music. The use of formats that support DRM is in an attempt to prevent copyright infringement of copyright protected materials, but methods exist to defeat most protection schemes, although such methods are considered illegal in many countries.


Encoding audio
The MPEG-1 standard does not include a precise specification for an MP3 encoder. The decoding algorithm and file format, as a contrast, are well defined. Implementers of the standard were supposed to devise their own algorithms suitable for removing parts of the information in the raw audio (or rather its MDCT representation in the frequency domain). During encoding 576 time domain samples are taken and are transformed to 576 frequency domain samples. If there is a transient 192 samples are taken instead of 576. This is done to limit the temporal spread of quantization noise accompanying the transient.

This is the domain of psychoacoustics: the study of subjective human perception of sounds.

As a result, there are many different MP3 encoders available, each producing files of differing quality. Comparisons are widely available, so it is easy for a prospective user of an encoder to research the best choice. It must be kept in mind that an encoder that is proficient at encoding at higher bitrates (such as LAME, which is in widespread use for encoding at higher bitrates) is not necessarily as good at other, lower bitrates.


Decoding audio
Decoding, on the other hand, is carefully defined in the standard. Most decoders are "bitstream compliant", meaning that the decompressed output they produce from a given MP3 file will be the same (within a specified degree of rounding tolerance) as the output specified mathematically in the ISO/IEC standard document. The MP3 file has a standard format which is a frame consisting of 384, 576, or 1152 samples (depends on MPEG version and layer) and all the frames have associated header information (32 bits) and side information (9, 17, or 32 bytes, depending on MPEG version and stereo/mono). The header and side information help the decoder to decode the associated Huffman encoded data correctly.

Therefore, for the most part, comparison of decoders is almost exclusively based on how computationally efficient they are (i.e., how much memory or CPU time they use in the decoding process).


Bit rate
The bit rate is variable for MP3 files. The general rule is that more information is included from the original sound file when a higher bit rate is used, and thus the higher the quality during playback. In the early days of MP3 encoding, a fixed bit rate was used for the entire file.

Bit rates available in MPEG-1 Layer 3 are 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kbit/s, and the available sampling frequencies are 32, 44.1 and 48 kHz. 44.1 kHz is almost always used (coincides with the sampling rate of compact discs), and 128 kbit/s has become the de facto "good enough" standard, although 192 kbit/s is becoming increasingly popular over peer-to-peer file sharing networks. MPEG-2 and the (unofficial) MPEG-2.5 include some additional bit rates: 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 kbit/s; while providing lower sampling frequencies (8, 11.025, 12, 16, 22.05 and 24 kHz).

Variable bit rates (VBR) are also possible. Audio in MP3 files is divided into frames, each of which has its own bitrate, so it is possible to change the bit rate dynamically as the file is encoded. This technique makes it possible to use more bits for parts of the sound with higher dynamics (more sound movement) and fewer bits for parts with lower dynamics, further increasing quality and decreasing storage space. For example, a portion composed of pure tones could be encoded at 48 kbit/s, taking up less space without any noticeable difference, while a portion played by a full symphony orchestra is encoded at 224 kbit/s to express it with greater fidelity. Although not originally implemented, many encoders now use this technique to greater or lesser extent.

Non-standard bitrates up to 640 kbit/s can be achieved with the LAME encoder and the --freeformat option, but few MP3 players can play those files. Gabriel Bouvigne, a principal developer of the LAME project, offered the following information about freeformat streams: [1]

"freeformat IS COMPLIANT with the mp3 standard. Decoders are required to be able to decode it up to 320kbps, but decoding higher bitrate freeformat streams is not mandatory.


Audio quality
Because MP3 is a lossy format, it is able to provide a number of different options for its "bit rate" — that is, the number of bits of encoded data that are used to represent each second of audio. Typically, rates chosen are between 128 and 320 kilobits per second. By contrast, uncompressed audio as stored on a compact disc has a bit rate of 1411.2 kbit/s (16 bits/sample × 44100 samples/second × 2 channels).

MP3 files encoded with a lower bit rate will generally play back at a lower quality. With too low a bit rate, "compression artifacts" (i.e., sounds that were not present in the original recording) may be audible in the reproduction. A good demonstration of compression artifacts is provided by the sound of applause: it is hard to compress because of its randomness and sharp attacks. Therefore compression artifacts can be heard as ringing or pre-echo.

As well as the bit rate of the encoded file, the quality of MP3 files depends on the quality of the encoder and the difficulty of the signal being encoded. As the MP3 standard allows quite a bit of freedom with encoding algorithms, different encoders may feature quite different quality, even when targeting similar bitrates. As an example, in a public collective test[2] (07/2003) featuring two different MP3 encoders at about 128kbps, one scored 3.66 on a 1-5 scale, while the other scored only 2.22.

Quality is heavily dependent on the choice of encoder and encoding parameters. While quality around 128kbps was somewhere between annoying and acceptable with older encoders, modern MP3 encoders can provide very good quality at those bitrates [3] (01/2006), not statistically different from quality provided by AAC, the technical successor of MP3. However, in 1998, MP3 at 128kbps was only providing quality equivalent to AAC-LC at 96kbps and MP2 at 192kbps [4].

The transparency threshold of MP3 can be estimated to be at about 128k with good encoders on typical music as evidenced by its strong performance in the above test, however some particularly difficult material can require 192k or higher. As with all lossy formats, some samples can not be encoded perfectly transparent to all users. Thus many users opt for 192k as a good trade off.

At lower bitrates, the quality of MP3 quickly degrades, and is far behind AAC quality at 32kbps, as demonstrated by a collective listening test (06/2004)[5].

It is also important to note that perceived quality can be influenced by listening environment (ambient noise), listener attention, and listener training.


File structure

Breakdown of an MP3 File's Structure An MP3 file is made up of multiple MP3 frames which consist of the MP3 header and the MP3 data. This sequence of frames is called an Elementary stream. Frames are independent items: one can cut the frames from a file and an MP3 player would be able to play it. The MP3 data is the actual audio payload. The MP3 header consists of a sync word which is used to identify the beginning of a valid frame. This is followed by a bit indicating that this is the MPEG standard and two bits that indicate that layer 3 is being used, hence MPEG-1 Audio Layer 3 or MP3. After this, the values will differ depending on the MP3 file. The range of values for each section of the header along with the specification of the header is defined by ISO/IEC 11172-3. Most MP3 files today contain ID3 metadata which precedes or follows the MP3 frames.

 

Design limitations
There are several limitations inherent to the MP3 format that cannot be overcome by using a better encoder.

Newer audio compression formats such as Vorbis and AAC no longer have these limitations.

In technical terms, MP3 is limited in the following ways:

Bitrate is limited to a maximum of 320 kbit/s (while some encoders can create higher bitrates, there is little-to-no support for these higher bitrate mp3s)
Time resolution can be too low for highly transient signals, causing some smearing of percussive sounds
Frequency resolution is limited by the small long block window size, decreasing coding efficiency
No scale factor band for frequencies above 15.5/15.8 kHz
Joint stereo is done on a frame-to-frame basis
Encoder/decoder overall delay is not defined, which means lack of official provision for gapless playback. However, some encoders such as LAME can attach additional metadata that will allow players that are aware of it to deliver gapless playback.
Nevertheless, a well-tuned MP3 encoder can perform competitively even with these restrictions.


ID3 and APEv2 tag
A "tag" in a compressed audio file, is a section of the file that contains metadata such as the title, artist, album, track number or other information about the file's contents.

As of 2006, the most widespread standard tag formats are ID3v1 and ID3v2, and the more recently introduced APEv2.

APEv2 was originally developed for the MPC file format (see the APEv2 specification). APEv2 can coexist with ID3 tags in the same file, but it can also be used by itself.

Tag editing functionality is often built-in to MP3 players and editors, but there also exist tag editors dedicated to the purpose.


Volume normalization
As compact discs and other various sources are recorded and mastered at different volumes, it is useful to store volume information about a file in the tag so that at playback time, the volume can be dynamically adjusted.

A few standards for encoding the gain of an MP3 file have been proposed. The idea is to normalize the average volume (not the volume peaks) of audio files, so that the volume does not change between consecutive tracks. This should not be confused with dynamic range compression (DRC) which is a form of normalization used in audio mastering.

The most popular and widely used solution for storing replay gain is known simply as "Replay Gain". Typically, the average volume and clipping information about audio track is stored in the metadata tag.

Thomson Consumer Electronics controls licensing of the MPEG-1/2 Layer 3 patents in many countries, including the United States, Japan, Canada and EU countries[6]. Thomson has been actively enforcing these patents. Due to different practices in different European countries when granting software patents under the European Patent Convention, it is unclear whether the patents would be upheld by national European courts.


In September 1998, the Fraunhofer Institute sent a letter to several developers of MP3 software stating that a license was required to "distribute and/or sell decoders and/or encoders". The letter claimed that unlicensed products "infringe the patent rights of Fraunhofer and THOMSON. To make, sell and/or distribute products using the [MPEG Layer-3] standard and thus our patents, you need to obtain a license under these patents from us." [citation needed]

These patent issues significantly slowed the development of unlicensed MP3 software [citation needed] and led to increased focus on creating and popularizing alternatives such as WMA and Ogg Vorbis. Microsoft, the makers of the Windows operating system, chose to move away from MP3 to their own proprietary Windows Media formats to avoid the licensing issues associated with the patents [citation needed]. Until the key patents expire, unlicensed encoders and players appear to be infringing articles in countries that recognize those patents.

In spite of the patent restrictions, the perpetuation of the MP3 format continues; the reasons for this appear to be the network effects caused by:

familiarity with the format,
the large quantity of music now available in the MP3 format,
the wide variety of existing software and hardware that takes advantage of the file format,
the lack of DRM restrictions, which makes MP3 files easy to edit, copy and distribute over networks,
the majority of home users not knowing or not caring about the patents controversy, which is in general irrelevant to their choice of the music format for personal use.
Additionally, patent holders declined to enforce license fees on open source decoders, allowing many free MP3 decoders to develop. [citation needed] Furthermore, while attempts have been made to discourage distribution of encoder binaries, Thomson has stated that individuals using free MP3 encoders are not required to pay fees. [citation needed] Thus while patent fees have been an issue for companies attempting to use MP3, they have not meaningfully impacted users, allowing the format to grow in popularity.

Sisvel S.p.A. [8] and its US subsidiary Audio MPEG, Inc. [9] previously sued Thomson for patent infringement on MP3 technology[10], but those disputes were resolved in November 2005 with Sisvel granting Thomson a license to their patents. Motorola also recently signed with Audio MPEG to license MP3-related patents. With Thomson and Sisvel both owning separate patents which they claim are needed by the codec, the legal status of MP3 remains unclear.

In September 2006 German officials seized MP3 players from SanDisk's booth at the IFA show in Berlin after an Italian patents firm won an injunction against the company in a dispute over licencing rights. The injunction was later reversed by a Berlin judge [11]; but that reversal was in turn blocked the same day by another judge from the same court, "bringing the Patent Wild West to Germany" in the words of one commentator. [12].


Ogg Vorbis from the Xiph.org Foundation, a patent free codec with free software implementations available.
MPEG-1/2 Audio Layer 2 (MP2), MP3's predecessor;
MPEG-4 AAC, MP3's successor, used by Apple's iTunes Music Store and iPod
MPC, also known as Musepack (formerly MP+), a derivative of MP2;
mp3PRO from Thomson Multimedia combining MP3 with SBR;
AC-3, used in Dolby Digital and DVD;
ATRAC, used in Sony's Minidisc;
Windows Media Audio (WMA) from Microsoft.
QDesign, used in QuickTime at low bitrates;
AMR-WB+ Enhanced Adaptive Multi Rate WideBand codec, optimized for cellular and other limited bandwidth use;
RealAudio from RealNetworks, frequently in use for streaming on websites;
Speex, a patent free codec based on CELP specifically designed for speech and VoIP, with free software implementations available.
mp3PRO, MP3, AAC, and MP2 are all members of the same technological family and depend on roughly similar psychoacoustic models. The Fraunhofer Gesellschaft owns many of the basic patents underlying these codecs, with Dolby Labs, Sony, Thomson Consumer Electronics, and AT&T holding other key patents.

There are also some lossless audio compression methods used on the Internet. While they are not similar to MP3, they are good examples of other compression schemes available. These include:

FLAC stands for 'Free Lossless Audio Codec'
Monkey's Audio
SHN, also known as Shorten
TTA
WavPack
Apple Lossless
Listening tests have attempted to find the best-quality lossy audio codecs at certain bitrates. At 128 kbit/s, Ogg Vorbis, AAC, MPC and WMA Pro tied for first place with LAME MP3 a little behind. At 64 kbit/s, AAC-HE and mp3pro performed marginally better than other codecs. At high bitrates (128 kbit/s+), most people do not hear significant differences. What is considered 'CD quality' is quite subjective.

Though proponents of newer codecs such as WMA and RealAudio have asserted that their respective algorithms can achieve CD quality at 64 kbit/s, listening tests have shown otherwise; however, the quality of these codecs at 64 kbit/s is definitely superior to MP3 at the same bitrate. The developers of the patent-free Ogg Vorbis codec claim that their algorithm surpasses MP3, RealAudio and WMA sound quality, and the listening tests mentioned above support that claim. Thomson claims that its mp3PRO codec achieves CD quality at 64 kbit/s, but listeners have reported that a 64 kbit/s mp3PRO file compares in quality to a 112 kbit/s MP3 file and does not come reasonably close to CD quality until about 80 kbit/s.

MP3, which was designed and tuned for use alongside MPEG-1/2 Video, generally performs poorly on monaural data at less than 48 kbit/s or in stereo at less than 80 kbit/s.

 

Hmmm........ i like analog even more now.

The ONLY REAL use for data compression is to appease the masses of morons that like lots of music ona small player. IE the ipod.


In this greater compaction of music we lose more and more of the MUSIC. Granted, most folks wont know the difference between 128 kb and 320kb, but some of us do. In the car asuio enviorment this can actualy be an advantage, what with higher nosie flooers, the lack of dynamic range can actually improve the listening experience. In the home audio/studio arena, the lack of dyanamics can rally make a set of spakers sound like a dog felched another dogs asss.

Lets be real here, the ONLY real limitation with lower bitrates is lack of dynamics, i have heard it with my own 2 ears. the frequency spectrum is there, the signal to noise ratio is still great, BUT, the dynamics are gone. They exist on the level of FM radio in my opinion.

I dont know, nor do i want to learn the ideas of dynamic compression.....UNLESS, it leads to the reopening of the dynamics in the playback. that would kick ass.


That is my opinion and i am sticking too it!