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> Quality & Bitrates
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This topic is intended to give torrentech users some guidelines about how to find and describe the quality of the audio files they want to share. It may not be totally accurate, this is because it's very simplified.
It is divided into three parts:
An explanation of the formats allowed on tT, an overview on some of the programs you can use to get that information, and a glossary of important terms.

There are links to more detailed information in the text, but basically all you need to know to post correctly on tT is here.
All the terms in green are explained in the glossary.

Important: reporting the bitrate of the files contained in a torrent as higher than it actually is harms Torrentech and will grant you a warning

  1. Explanation of the formats allowed on tT:

    On torrentech, you are allowed to post in one of these formats: MP3, FLAC , AAC and OGG Vorbis.

    Important: the extension of your files doesn't allways match the name of the codec used. For example, AAC files often come in containers with a .mp4 or a .m4a extension. Allways use one of the programs in section 2 before you post a release.

    If you have files in other lossless formats (APE, ALAC, Wavpack, etc...) you have to convert them to FLAC to be able to upload them. For this purpose you can use dBPowerAmp.

    • FLAC
      FLAC is lossless audio compression, please pay attention to enter the correct bitrate of the release. If you want information about this format you can read this: FLAC.

    • MP3
      MP3 is lossy. This means it encodes the audio using a determined bitrate that is insufficient to store all the information. The quality of the audio file after this conversion depends on the bitrate chosen. On higher bitrates, less information is lost during the conversion, and the quality of the resulting audio file is higher.

      MP3 has two types of bitrate: VBR and CBR

      • VBR stands for "Variable BitRate". The encoder "listens" to each fragment of audio in the song and determines how many bits it needs to achieve a certain quality measure. This means that each part of the song is encoded with a different number of bits.
        To define the quality goal, VBR MP3 has many presets going from V0 to V9, V0 being the one with the highest mean bitrate, and thus the least loss of information.
        On tT you are allowed to post from V0 to V4. The following table shows the features of each one of them:



        There is a setting in VBR called ABR (Average BitRate) In this setting, the variations in bitrate are smaller, and depend on a target bitrate. This type of encoding is not very common. It is halfway between VBR and CBR.
        You are allowed to post ABR MP3 in these bitrates:
        • 128 Kbps
        • 160 Kbps
        • 192 Kbps
        • 224 Kbps
        • 256 Kbps
        • 320 Kbps

        Note that if an ABR release does not exactly fit into one of these categories, you should pick the closest one.

        For more info: ABR, VBR
        .
      • CBR stands for "Constant BitRate". The bitrate remains constant through the entire length of the song. The quality of the final file is determined by that bitrate. Of course, the higher the number, the better the quality. On tT you are allowed to post CBR MP3 in these bitrates:
        • 128 Kbps
        • 160 Kbps
        • 192 Kbps
        • 224 Kbps
        • 256 Kbps
        • 320 Kbps ([alt-] preset insane)

        For more info : CBR

      For more info about MP3 read this: MP3 or LAME

    • AAC
      AAC stands for "Advanced Audio Coding". It's also a lossy format. AAC audio is normally stored in a .mp4 or a .m4a file.
      It is designed to improve upon and replace MP3 as the defacto Audio Encoding standard. It supports VBR and CBR encoding.
      AAC's encoding settings are very complex. On tT we just require you to select the bitrate from the list that is the closest to the bitrate of your files (don't post AAC with bitrates lower than 128 Kbps):
      • 128 Kbps
      • 160 Kbps
      • 192 Kbps
      • 224 Kbps
      • 256 Kbps
      • 320 Kbps


      For more info: AAC, AAC FAQ

    • Ogg Vorbis
      Vorbis is a lossy codec, usually found in the Ogg container, producing Ogg Vorbis files. It uses a variable bitrate algorithm, with different presets numbered from q-2 to q10, being q10 the one with the highest mean bitrate. The presets that are alowed can be seen on the following table:



      For more info: Vorbis


  2. How to know what quality your files are:

    If you didn't encode your music yourself and you're sharing from another source, or you've forgotten how a release was encoded, and aren't sure of how to determine what the bitrate or encoder was, there are many ways to find out.
    If you want to learn to encode your own music, read this: Step by step guide to secure CD ripping

    In the following list you will find some programs that can be used to determine the quality of your audio files:

    • foobar2000: http://foobar2000.org

      If you use foobar2000 to play your music, it also works wonderfully for identifying LAME-encoded MP3s. Right-click on a file (or a selected group of files) and select Properties. Select the Properties tab and it will look something like this:



      Note that foobar2000 will only show LAME-encoded MP3 profiles. This should not be a major problem, as the majority of releases are provided this way. Releases using other encoders should be considered as potentially sub-standard.
      .
    • Mr QuestionMan: http://www.burrrn.net/?page_id=5

      This program is very easy, almost point and click to use. When you use this program and browse to a folder of music you'll be presented with some information on all the files within the directory.

      In the following example you can see MrQuestionMan at work. To use it just select the directory where your files are in the list at the left side. It will show you a list of the files inside that directory and provide information about them.



      In this example you see it writes MPEG 1 Layer III LAME 3.90 [alt-]preset standard. You can see in the bitrate column that it is a VBR, as all the files have different (and allmost random) values.
      • MPEG 1 Layer III is just a hard way to say MP3
      • LAME 3.90 is the name and version of the encoder used.
      • And now the important part. If you look at the table of VBR settings in section 1 of this post, you will notice that [alt-]preset standard corresponds to a VBR MP3 with the preset V2. You now know what you want.

      .
    • Audio Identifier: http://www.download.com/Audio-Identifier/3...4-10703772.html

      This is another program that will provide you with the same information. The only difference here is that the encoder has it's own column separate from encoding options. Also, if the encoder used is LAME, you can right click on any file and view the LAME tag. This is a place in the file that LAME uses to store some more specs about the rip. Sometimes it's the only way to get the right information.



      In this example it wouldn't be necessary because it is obvious that it is a VBR V3 MP3. But be carefull, sometimes audioidentifier writes a V# setting in the "encoder options" column when it is a CBR. Allways look at the bitrate column first!!
      .
    • MediaInfo: http://mediainfo.sourceforge.net/en/Download (download the GUI version)

      This program is a bit more complicated, but it's the only option I have found for MAC OS.



      You have to select track by track. If you click on the information panel, you get the yellow alt-text. At the bottom are the encoding options. In this case MP3 VBR V0.
      .
    • Tag&Rename: http://www.download.com/Tag-Rename/3000-21...cdlpid=10810533

      This program's options seem a lot more complicated, as they are. Beyond providing us with information about the release, it also allows you to do other wonderful music related tasks, like mass editing tags and more.


  3. Glossary and important terms:
    • bitrate: it's the number of bits used per second in an audio file or a comunication. 320 Kbps for example means 320.000 bits per second
    • container: It's the file where the audio is stored. It also contains the tags and other information about the audio.
    • encoder/codec: it's the program that transforms the audio into a specific format, like MP3 or FLAC. Codec stands for coder/decoder.
    • lossless: an audio file that has been encoded in a way it can be encoded back to his original form without any loss.
    • lossy: an audio file that has been encoded in a way it can't be encoded back to it's original form without loss. Lossy encoders usually try to remove the parts of the audio humans can't hear, or are less important for the human ear.

      How lossy audio quality works

      Lossy codecs, such as MP3, work by throwing out parts of music that we cannot hear. The ultimate goal of a lossy codec is transparency, which means that the lossy file is indistinguishable from the lossless. If a file is transparent, increasing the bitrate does not improve quality. Using LAME, transparency can be achieved as low (or lower than) V5 for average listeners with average equipment.

      Transparency depends on a few things. First, and most important, is the material being encoded. Some samples cannot be transparently encoded with MP3. On the other hand, there are some samples that achieve transparency at bitrates below 100kbps. There are some samples where improving your listening equipment will improve your ability to hear when the files are not transparent. However, overall, most audio files can be transparently encoded with LAME at V2 and no amount of expensive audio equipment will allow you to discern it from lossless using just your ears.

      Note that these comments only hold for listening. By looking at a spectrogram, it is quite easy to spot which files were encoded using MP3!
    • transcode: transcoding is using a file encoded in a determined encoding scheme as the source for another encoder. If the source file is on a lossless format, there is no problem, but if the source file is lossy, the different quality losses of both encoders sum up, giving the target file a quality that is worse than expected. Lossy to lossy and lossy to lossless transcodes are strictly forbidden. More info: Transcoding
    • mutt rip: A mutt rip is an album which different tracks come from different sources. These where common with napster, limewire and such when you had to download audio files individually. Mutt rips are also forbidden. More info: Mutt Rips
Lo-Fi Version Time is now: 18.01.2019 - 15:52