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this article is reprinted from the Fraunhofer website
Frequently Asked Questions
- What´s MPEG
- MPEG is the "Moving Picture Experts Group", working under the joint direction of
the International Standards Organization (ISO) and the International
Electro-Technical Commission (IEC). This group works on standards for
the coding of moving pictures and audio. MPEG has created its own homepage,
providing information on the what, where, when and how of the standards.
- Differences between MPEG-1 and MPEG-2 audio
- MPEG-1 and MPEG-2 use the same family
of audio codecs, Layer-1, -2 and -3. The new audio features of MPEG-2
are a "low sample rate extension" to address very low bitrate applications
with limited bandwidth requirements (the new sampling frequencies are
16, 22.05 or 24 kHz, the bitrates extend down to 8 kbps), and a "multichannel
extension" to address surround sound applications with up to 5 main
audio channels (left, center, right, left surround, right surround)
and optionally 1 extra "low frequency enhancement (LFE)" channel for
subwoofer signals; in addition, a "multilingual extension" allows the
inclusion of up to 7 more audio channels.
- What´s MPEG 2.5
- The MPEG-2 standard allows bitrates as low as 8 kbps, for the low sample rate extension. At such a low bitrate, the useful audio bandwidth has to be limited anyway, e.g. to 3 kHz. Therefore, the actual sample rate could be reduced, e.g. to 8 kHz. The lower the sample rate, the better the frequency resolution, the worse the time resolution, and the better the ratio between control information and audio payload inside the bitstream format. As the MPEG-2 standard defines 16 kHz as lowest sample rate, we introduced a further extension, again dividing the low sample rates of MPEG-2 by 2, i.e. we introduced 8, 11.025, and 12 kHz - and we named this extension to the extension "MPEG 2.5". "Layer-3" performs significantly better with 8 kbps @ 8 kHz or 16 kbps @ 11 kHz than with 8 or 16 kbps @ 16 kHz.
- The MPEG-2 standard allows bitrates as low as 8 kbps, for the low sample rate extension. At such a low bitrate, the useful audio bandwidth has to be limited anyway, e.g. to 3 kHz. Therefore, the actual sample rate could be reduced, e.g. to 8 kHz. The lower the sample rate, the better the frequency resolution, the worse the time resolution, and the better the ratio between control information and audio payload inside the bitstream format. As the MPEG-2 standard defines 16 kHz as lowest sample rate, we introduced a further extension, again dividing the low sample rates of MPEG-2 by 2, i.e. we introduced 8, 11.025, and 12 kHz - and we named this extension to the extension "MPEG 2.5". "Layer-3" performs significantly better with 8 kbps @ 8 kHz or 16 kbps @ 11 kHz than with 8 or 16 kbps @ 16 kHz.
- MPEG-3 and Layer-3
- Layer-3 is a powerful audio coding scheme which certainly is part of the MPEG standard. Layer-3 is defined within the audio part of both existing international standards, MPEG-1 and MPEG-2.
But:There is no MPEG 3 defined.
- Layer-3 is a powerful audio coding scheme which certainly is part of the MPEG standard. Layer-3 is defined within the audio part of both existing international standards, MPEG-1 and MPEG-2.
- Layer-1, Layer-2, Layer-3
- MPEG describes the compression of audio signals using high performance perceptual coding schemes. It specifies a family of three audio coding schemes, simply called Layer-1, Layer-2, and Layer-3. From Layer-1 to Layer-3, encoder complexity and performance (sound quality per bitrate) are increasing. The three codecs are compatible in a hierarchical way, i.e. a Layer-N decoder may be able to decode bitstream data encoded in Layer-N and all Layers below N (e.g., a Layer-3 decoder may accept Layer-1,-2,-3, whereas a Layer-2 decoder may accept only Layer-1 and -2.)
Each audio Layer extends the features of the Layer with the lower number. The simplest form is Layer-1. It has been designed mainly for the DCC (Digital Compact Cassette), where it is used at 384 kbps (called "PASC"). Layer-2 has been designed as a trade-off between complexity and performance. It achieves a good sound quality at bitrates down to 192 kbps. Below, sound quality suffers. Layer-3 has been designed for low bitrates right from the start. It adds a number of "advanced features" to Layer-2: the frequency resolution is 18 times higher, which allows a Layer-3 encoder to adapt the quantization noise much better to the masking threshold only Layer-3 uses entropy coding (like MPEG video) to further reduce redundancy only Layer-3 uses a bit reservoir (like MPEG video) to suppress artifacts in critical moments and Layer-3 may use more advanced joint-stereo coding methods.
All Layers use the same basic structure. The coding scheme can be described as "perceptual noise shaping" or "perceptual subband / transform coding". The encoder analyzes the spectral components of the audio signal by calculating a filterbank (transform) and applies a psychoacoustics model to estimate the just noticeable noise-level. In its quantization and coding stage, the encoder tries to allocate the available number of data bits in a way to meet both the bitrate and masking requirements. The decoder is much less complex. Its only task is to synthesize an audio signal out of the coded spectral components.
All Layers use the same analysis filterbank (polyphase with 32 subbands). Layer-3 adds a MDCT transform to increase the frequency resolution. All Layers use the same "header information" in their bitstream, to support the hierarchical structure of the standard.
All Layers have a similar sensitivity to biterrors. They use a bitstream structure that contains parts that are more sensitive to biterrors ("header", "bit allocation", "scalefactors", "side information") and parts that are less sensitive ("data of spectral components"). All Layers support the insertion of programm-associated information ("ancillary data") into their audio data bitstream.
All Layers may use 32, 44.1 or 48 kHz sampling frequency.
All Layers are allowed to work with similar bitrates:
- Layer-1: from 32 kbps to 448 kbps
- Layer-2: from 32 kbps to 384 kbps
- Layer-3: from 32 kbps to 320 kbps
- MPEG describes the compression of audio signals using high performance perceptual coding schemes. It specifies a family of three audio coding schemes, simply called Layer-1, Layer-2, and Layer-3. From Layer-1 to Layer-3, encoder complexity and performance (sound quality per bitrate) are increasing. The three codecs are compatible in a hierarchical way, i.e. a Layer-N decoder may be able to decode bitstream data encoded in Layer-N and all Layers below N (e.g., a Layer-3 decoder may accept Layer-1,-2,-3, whereas a Layer-2 decoder may accept only Layer-1 and -2.)
- What is MPEG-2 AAC?
- MPEG-2 AAC (Advanced Audio Coding) is the latest MPEG standard on perceptual audio coding. It has been standardized as ISO 13818-7 in 04/1997. It is the consequent extension of existing standards towards optimum coding efficiency. AAC is not backwards compatible towards MPEG Audio Layers-1,2,3 and thus can incorporate the latest research results without any compatibility restrictions. Due to that reason, AAC was called NBC (non backwards compatible) in early development stages.
- MPEG-2 AAC (Advanced Audio Coding) is the latest MPEG standard on perceptual audio coding. It has been standardized as ISO 13818-7 in 04/1997. It is the consequent extension of existing standards towards optimum coding efficiency. AAC is not backwards compatible towards MPEG Audio Layers-1,2,3 and thus can incorporate the latest research results without any compatibility restrictions. Due to that reason, AAC was called NBC (non backwards compatible) in early development stages.
- Who developed MPEG-2 AAC?
- MPEG-2 AAC was developed inside MPEG. During the standardization process, numerous companies contributed to the standard. Among them were the Fraunhofer Institute for Integrated Circuits IIS, AT&T, Dolby, Sony, the University of Hannover and NEC. Fraunhofer IIS-A developed main parts of the encoder and held responsible for the integration of all contributions and audio quality optimization.
What are the main differences between
MPEG-2 AAC and MPEG Layer-3?- AAC uses a different type of transform and incorporates a lot of additional features to enhance coding efficiency like Temporal Noise Shaping and Prediction. Additionally, AAC uses a very flexible entropy coding kernel to transmit coded spectral data. All these enhancements plus the careful tuning done by Fraunhofer IIS-A provide you with the best audio coding scheme you can get today
- MPEG-2 AAC was developed inside MPEG. During the standardization process, numerous companies contributed to the standard. Among them were the Fraunhofer Institute for Integrated Circuits IIS, AT&T, Dolby, Sony, the University of Hannover and NEC. Fraunhofer IIS-A developed main parts of the encoder and held responsible for the integration of all contributions and audio quality optimization.
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