MP3 encoder


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MP3 encoder

Mp3 Encoder
Mp3 Encoder

1. MP3 Encoder FAQ

Mp3 Encoder
Mp3 Encoder

: what is an MP3 encoder?
An MP3 encoder is a piece of software that uses the MP3 codec algorithm (compression/decompression) to create mp3 files. Most encoders only convert
a WAV file to an MP3 file, although many can convert other formats such as WMA, Real Audio, Ogg, etc.

There are only a few standalone encoders, and a lot of software also only uses 4 main encoding engines, largely due to
to Fraunhofer Gesellschaft patents and various companies helping with ISO sources. Although no company owns the license, the
Developers must pay expensive license fees no matter what proprietary MP3 encoder they use. Major MP3 encoding engines include: LAME (
non-ISO source), BladeEnc, Fraunhofer, and Real Networks’ Xing encoder.

– How does the MP3 encoder work?
The core technology under MPEG-Layer 3 is included in the MP3 encoder. The decoding process uses a series of algorithms and rules to compress audio.
The encoder also detect sounds that occur at the same time
and they try to rule out any that might be “masked” or “inaudible” by other sounds.

– What is a good MP3 encoder?
Xing is the fastest encoder in terms of speed, but the worst in quality. For smaller file sizes, Fraunhofer FastEnc
offers the best quality. LAME is a very good encoder, and one version is faster than the previous one, BladeEnc
it is the best quality for large files, but very slow.

2. Dissection of MP3 files
In addition to proficiency in using the basic features of the MP3 encoder, ordinary users do not need to know how the internal structure of the MP3 file is encoded, just like the situation when
face JPEG or DOC files. Out of morbid curiosity, here’s an X-ray view of an MP3 file:

– Box header
As mentioned above, MP3 files are made up of thousands of “frame frames”, each frame containing a part (second part) of valuable audio data.
for the decoder to reconstruct the audio data. The first part above is the box header. (Frame Header), which consists of 32-bit metadata related to the
later data, see the figure below. The MP3 header begins with an 11-bit “sync timing” block, which allows the player to seek and lock the first
legal framework available, which is useful in MP3 streaming, which can quickly move or jump ID3 from the playback source block to a normal one.
position . However, simply detecting synchronized blocks is theoretically not enough, so it is necessary to check the header.

– transmission lock
MP3 was originally designed for broadcast, and as a result it became important that the MP3 receiver could be synchronized with the signal at any part of the broadcast,
so the frame header is placed at the beginning of any frame transmission, so when an MP3 receiver “tunes” to a data stream, it picks up the
signal instantly and you can play it immediately. Interestingly, this fact makes it possible to cut MPEG files into small segments, each of which can be played independently. But unfortunately
not possible in 3-layer (MP3) files, where frames often depend on other frames, so you can’t just
Edit .

– Frames per second
Just as the movie industry has a standard for the number of frames per second in film to ensure proper viewing on any projector,
A similar standard is used in the MP3 standard, regardless of the file’s bitrate, MPEG-1 A frame in the file is 26 ms, approximately 38 fps frames per second. If the bit rate
is , the frame size is correspondingly larger, and vice versa. Also, the number of samples contained in an MP3 frame is constant, 1152 samples per frame.

The total size of any given frame can be calculated with the following formula:

FrameSize = 144 * BitRate / (SampleRate + Padding).


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Audio quality: Bitrate in MP3 files

In many cases the term Bitrate is used, which is the bit rate per second that a multimedia file (Audio or Video) has. Currently the MP3 music format is one of the most widespread (Although there are currently other more current formats such as OGG Vorbis, AAC, Flac, Monkey Audio, …) however the audio quality is variable, this is due to the characteristics with which the MP3 in question has been compressed, including:

Mode: It can be of two types mainly:
Mono: With a single channel (The right and left channel go together, not separated which gives worse audio quality).
Stereo: Two channels (Right and Left, improve audio quality).

Sampling frequency: Audio CDs use 44,100 Hz (22,050 Hz per channel), although there are higher frequencies such as 48,000 Hz used in DVDs and lower, the higher the frequency, the higher the quality.

Bits: Audio CDs have 16 Bits (Although MP3 can be compressed at a lower quality such as 8 Bits).

Bitrate (Bit Rate per second): Audio CDs have about 1,400 Kbps (44100 Hz * 16 Bits * 2 channels), meaning that an Audio CD would have a bitrate of 1,400 Kbps (In MP3 format the maximum Bitrate is 320 Kbps, however, it is assumed that an MP3 with a 128 Kbps Bitrate has a quality similar to CD, although in many cases to achieve a quality similar to CD it is necessary to use a Bitrate of 192 Kbps, and to obtain CD quality it is necessary use 256 Kbps or 320 Kbps).

Some of the most common Bitrates are:
8 Kbps Mono: Telephone Sound.
16 Kbps Mono: Better quality than shortwave.
32 Kbps Mono: Better quality than AM.
64 Kbps Stereo: Better quality than FM.
112 – 128 Kbps: Quality close to CD.
160 Kbps: Quality closer to CD.
192 Kbps: Virtually CD quality.
256 Kbps: Quality CD practically undisputed from an original CD.
320 Kbps: CD quality.

Coding method: It can be of two types:
VBR (Variable Bit Rate, Bit Rate Variable): Encodes the file in MP3 with a variable Bitrate.
CBR (Constant Bit Rate, Constant Bit Rate): Encodes the MP3 file with a fixed Bitrate.
In addition, another factor that influences the encoding of the MP3 file is the CODEC (Encoder-Decoder) used, one of the most common and the best result is LAME (Lame Ain’t an MP3 Encoder) which is also free.
One point to keep in mind is that if we recompress an MP3 file that originally has a 128 Kbps bitrate and convert them to 192 Kbps for example, audio quality is not really gained because the MP3 format has some quality loss (MP3 is a loss algorithm, also called lossy). which has occurred when converting the original file (Ex: CD Audio or a 320 Kbps MP3 to a 128 Kbps MP3) so this recompression does not make much sense since we will not gain in audio quality (As they say where there is no one can not get) and the only thing we will achieve in any case is to increase the initial size of the file.
The opposite case (Recompress a 320 Kbps MP3 file for example at 192 Kbps) if it makes some sense because in this case although we lose some audio quality we reduce the weight (Kilobytes or Megabytes) of each MP3 file somewhat.
In conclusion, it can be said that if we need to encode / compress an MP3 file with good quality, the “ideal” would be to do so:
To be able to start from an Audio CD, although an MP3 at 320 or 256 Kbps could also be valid for a recompression of the file.
In stereo mode (With two channels, right and left).
With at least 44100 Khz sampling rate and 16 Bits.
With a minimum bitrate of 192 Kbps or at most 256 Kbps (Using 320 Kbps would give higher quality but also increase the file size considerably).