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Audio Codecs
Speech Codecs
Video Codecs
Multimedia Frameworks
Audio Processing Subsystem
 
 
Audio Codecs

The following is the list of Audio Decoders and Encoders with most of them completely developed and few of them under advanced stages of development, currently ported to two RISC processor families, namely ARC and ARM.  With our strong algorithm development skills and good flair to work on multiple processors, our team can quickly port these solutions on other DSP and RISC processors.

 
MP3 Decoder
 
MP3 is the popular short form of MPEG-1 Layer 3. MPEG-1 supports 3 layers. Layer-1 is the simplest and best suited for bit-rates above 128 kbps per channel. Layer-2 has an intermediate complexity and is targeted for bit-rates around 128 kbps per channel. Layer-3 is the most complex and offers the best audio quality, particularly for bit-rates around 64 kbps per channel. All the three layers support sampling frequencies of 32, 44.1 and 48 kHz. MPEG-2 BC provides for an extension of MPEG-1 towards sampling rates for low bit-rate applications.

Support for 16, 22.05 and 24 kHz is provided. Bit-rates from 32 to 256 kbps for layer 1 and 8 to 160 kbps for layers 2 and 3 are supported. It also provides a backward compatible multi-channel extension to MPEG-1. Incube MP3 decoder supports MPEG-1 layers 1,2 and 3 and MPEG-2 BC. They also support the low bit-rate extension MPEG-2.5. Data Sheet

 
MP3 Encoder
 
Thank your for your interest in this solution. The chosen solution is under development. Please revert to us later.
 
AAC-LC Decoder
 
AAC is designed as the successor to MP3. It achieves better sound quality than MP3 at many bit-rates, and is standardized by ISO and IEC in MPEG-2 and MPEG-4 specifications. AAC was first specified in the standard MPEG-2 Part 7 as a new part in MPEG-2 family of international standards. It provides a high audio scheme for 1 to 48 channels at sampling rates of 8 to 96 kHz. AAC-LC (low-complexity) is one of the three profiles of AAC provided by the MPEG-2 AAC standard. The AAC of MPEG-2 was later updated in MPEG-4 Part 3, with the notable addition of Perceptual Noise Substitution (PNS).

Incube AAC-LC decoder supports the MPEG-2 AAC-LC profile along with the additional tools provided by MPEG-4 standard. Data Sheet

 
AAC-LC Encoder
 
AAC is designed as the successor to MP3. It achieves better sound quality than MP3 at many bit-rates, and is standardized by ISO and IEC in MPEG-2 and MPEG-4 specifications. AAC was first specified in the standard MPEG-2 Part 7 as a new part in MPEG-2 family of international standards. It provides a high audio scheme for 1 to 48 channels at sampling rates of 8 to 96 kHz. AAC-LC (low-complexity) is one of the three profiles of AAC provided by the MPEG-2 AAC standard. The AAC of MPEG-2 was later updated in MPEG-4 Part 3, with the notable addition of Perceptual Noise Substitution (PNS). Data Sheet
 
AAC-Plus Decoder
 
AAC-Plus (AAC+) or HE-AAC is an extension of AAC-LC optimized for low-bitrate applications such as streaming audio. It uses Spectral Band Replication to enhance the compression efficiency in the frequency domain. HE-AAC v2 couples SBR with Parametric Stereo to enhance the compression efficiency of stereo signals. AAC+ v2 is also standardized by ETSI.

Incube AAC-Plus decoder supports SBR and Parametric Stereo, and has been highly optimized to be used even in power-constrained applications. Data Sheet

 
SBC Decoder
 
SBC is an audio coding system specifically designed for Bluetooth audio and video applications to obtain high-quality audio at moderate bit-rates with low computational complexity. SBC is implemented according to Bluetooth Advanced Audio Distribution Profile (A2DP) specifications as a mandatory codec for wireless applications. Apart from being computationally well suited for wireless (typically handheld) applications, SBC also has to capacity to adjust to changing bandwidths as is typical for any wireless connection.

Incube SBC decoder is very compact and efficient implementation of the SBC decoder. It provides flexibility to the end-user to tradeoff between complexity and performance. Data Sheet

 
FLAC Decoder
 
FLAC stands for Free Lossless Audio Codec. As the name indicates, it is a lossless audio codec, and is one of the widely used formats among lossless audio codecs. It is a low complex audio codec and the decoder real-time performance is achievable on even modest hardware. It is highly error-resistant, as each frame can be independently decoded, without depending on information from the previous or the following frame. FLAC typically is used with Ogg format. In addition it also supports a native container format.

The Incube FLAC decoder is a flexible and complete implementation of FLAC and is highly optimized in terms of both the memory and MHz. Data Sheet

 
Ogg-Vorbis Decoder
 
Ogg-Vorbis consists of two parts – Ogg container and Vorbis decoder. Though these two are theoretically independent, most implementations, including the reference implementation tightly couple these two parts. Vorbis is a fully open, non-proprietary, patent-and-royalty-free, general purpose compressed audio format for mid and high quality (8-48kHz, 16+bits, polyphonic) audio and music at fixed and variable bit-rates from 16 to 128 kbps/channel. Because of its “free” nature, Vorbis has gained quite a bit of popularity, support and adoption.

The Incube implementation of Vorbis clearly separates out the boundary between Ogg parsing and Vorbis decoding, thus making the Vorbis implementation generic enough to be used with any container format, if the need be such. The implementation has been carefully designed to provide sufficient flexibility for use in various types of applications. Data Sheet

 
Monkey’s Audio Decoder
 
Monkey’s Audio is a lossless audio compression format. Monkey’s Audio files use the file name extensions .ape for audio and .apl for track metadata. Monkey’s audio supports various levels of compression, by employing different types of filters internally. As the compression level increases, the complexity of encoding as well as decoding goes up. Data Sheet
 

To accommodate decoding of these streams from a multiplexed container format, we have demultiplexer implementations for the following formats.

 
DRA Decoder
 
DRA (Dynamic Resolution Adaptation) is a high quality multi-channel lossy audio compression algorithm, designed for low decoder complexity and is considered as equivalent to Dolby and DTS compression algorithms in quality and is as one of the optional formats for Blue-ray audio. It varies its temporal-frequency resolutions depending on the quasi-stationary or transient nature of the signal in a given window.

Incube team has ported DRA decoder on ARC XY and non-XY platforms. Incube 's implementation of the DRA decoder supports all the features of the standard (sampling rates of 8-192 kH, multiple channels up to 7.1 with down-mixing facility etc.,). In addition this implementation has features such as interlaced and non interlaced modes and 16, 24 and 32 bit-depths required for blue-ray. With this extensive working experience on DRA algorithm and Incube's team can port same on any platform.
 
WMA Encoder
 
The WMA system uses the waveform coding concepts of subband/transform coding, quantization, and variable length coding together with the models of properties of human auditory system to achieve the highest quality of audio even at low data rates. It employs a form of noise coding (frequency smearing) and line spectral pairs to achieve very low bit-rates.

Incube's team has ported the same on ARC XY-platform with all its standard features (sampling rates 8-32 kHz, bit rates 8 – 192 kbps, mono and stereo channels). Our team has extensive hands-on experience with this algorithm and capable of tuning the same to achieve lowest MHz requirement without trading of quality. Incube's implementation also has feature to change, in run-time, some of the parameters (such as bit-rate, sampling frequency, number of channels etc.,) to adjust the to the varying bandwidth availability.
 
HE-AAC Encoder
 
High Efficiency Advanced Audio Coding (HE-AAC), or MPEG-4 Audio, is a lossy audio compression algorithm used in applications such as digital radio. The version-2 profile of this algorithm (HE-AAC v2) is an extension of low complexity AAC with Spectral Band Replication (SBR) technique to effectively compress high frequencies and Parametric Stereo (PS) tool to achieve good compression of stereo signals at high quality.

Our team has in depth understanding of this algorithm,, in particular from the point of view of data flow and resource requirement, and porting experience in extremely memory resource constrained environment, enabling us to quickly port the algorithm on any platform.
 
Audio Demultiplexers:
Ogg
 
Ogg is a free, open-standard container format maintained by Xiph.Org foundation – the same foundation that maintains Vorbis and FLAC. Ogg is a generic file format that can multiplex a number of separate independent codecs for audio, video, text and metadata. Though Ogg can embed any audio/video formats, it was originally intended and is usually used with free codecs like Vorbis, Theora, FLAC, Speex, Writ, CMML, Annodex etc.

The Incube implementation of the Ogg demultiplexier is very flexible and provides for fast seeking and good error robustness. Data Sheet

 
Matroska
 
Matroska is an extensible open standard Audio/Video container maintained by matroska.org foundation. Matroska is usually found as .mkv files (matroska video) and .mka files (matroska audio). It is based on EBML (Extensible Binary Meta Language) which gives significant advantages in terms of future format extensibility, without breaking file support in old parsers. Matroska supports all advanced codecs. Data Sheet
 
Audio Processing:
Sample Rate Converter (SRC)
 
SRC is an algorithm used to convert an audio signal from one sample rate to another. SRC is an essential component in various audio systems today as applications use different sampling rates. Incube's Sample Rate Converter is designed to convert audio samples from any sample rate to any other sample rate. Algorithms with different quality vs. performance tradeoff are provided to allow the optimal implementation to be used in any given application.
Data Sheet
 
Parametric Equalizer
 
Equalizer is a software that enables the user to control gain in different frequency bands in an audio system. It is generally used to improve the fidelity of sound, to emphasize certain instruments or to remove undesired noises. The Incube's multiband parametric equalizer allows audio signals to be filtered with a bank of band-pass filters. The number of bands, the position and frequency range of the bands and the gain applied on each of them is configurable. The solution is very well optimized for embedded platforms by use of fast algorithms and effective optimization in assembly language.
Data Sheet
 
Value Added Features:
Our team is experienced in developing features such as
  • Developing codecs optimized for multiple platforms (for example ARM Cortex A8 with and without NEON SIMD processing and ARC platforms with and without XY-memory) with compile time selection.
  • Providing ability to encoders for run-time switching of parameters (such as bit-rate, sampling frequency, channel number etc.,) to suite varying channel bandwidth conditions.
  • Trick play in demultiplexers.
 
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