Audio Algorithms
AAC-LC / AAC-LD / MPEG4-AAC Encode / MPEG4-AAC Decode / MP3 Decode / WMA Decode
Features List
- C-callable API for initialization, decoding
- Multi-channel
- Optimized implementation
- eXpressDSP Digital Media compliant
AAC LC (Low Complexity) Encode / Decode Audio Algorithm
Sampling Rate: 8 khz – 96 khz
Bit Rate: 16 – 576 kbps
FEATURES
- Encoding and decoding of AAC-LC (Low Complexity) bit-streams
- MPEG4 AAC Low Complexity (LC) object type implementations supported
- MPEG2 AAC Low Complexity (LC) object type implementations supported
- Decoding of mono and stereo streams supported
- Mono, stereo, and dual mono input file supported
- RAW data input format supported
- C callable API for initialization, decoding
- Full accuracy with ISO/IEC 13818 and ISO/IEC 14496 audio standards
- Multi-channel, Reentrant implementation
- Tested on a variety of AAC bit streams
- 16/24 bit PCM output / High sound quality
- Only AAC-LC output format supported
Availability
| Platforms |
| Texas Instruments – TI TMS320C6000 C64x, C64x+, C674x |
ADT AAC LC is available on the above Platforms: Other configurations are available upon request.
DESCRIPTION
The AAC-LC (Advanced Audio Coding – Low Complexity) is a natural audio coding algorithm that can handle 48 channels and sample rates up to 96kHz. This coding technique uses a perceptual filter bank, a sophisticated masking model, noise-shaping techniques, and channel coupling. It provides the highest possible quality at smaller bit-rates. AAC is an audio data compression format. The encoder/decoder is an implementation of the MPEG-2 (ISO/IEC 13818-3) and MPEG-4 (ISO/IEC 14496-3) standard. AAC provides both a higher quality and superior performance than MP3 at the same bit rate. It supports the coding of multi-channel audio, with up to 48 main channels and 16 low-frequency channels. AAC offers multiple profiles/object types to meet the requirements of a wide range of applications making it one of the most popular audio compression standards across wide spectrum of application ranging from portable player, cell phones, music systems, to the internet.
AAC LC (Low Complexity) Encode / Decode Audio Algorithm
Sampling Rate: 480 samples and 512 samples per frame supported
Delay: 20 ms
FEATURES
- Suitable for all kinds of audio signals including speech and music
- Audio quality better than ITU-T G.722/G.722.1-C, G.729.1 at the same bit rate
- Audio quality better than mp3 at the same bit rate
- Algorithmic delay of only 20 ms
- Delay of real-time DSP implementation down to 30 ms
- Multi-channel support
- Large range of bit rates possible
- Computational and storage complexity comparable to MPEG-4 AAC-LC
- 480 samples and 512 samples per frame supported
- Error resilient AAC-LD audio object type
- Decoding of mono and stereo streams supported
Availability
| Platforms |
| Texas Instruments – TI TMS320C6000 C64x, C64x+, C674x |
ADT AAC LD is available on the above Platforms: Other configurations are available upon request.
SPECIFICATIONS
AAC-LD C64x
CPU Utilization & Memory RequirementsAll Memory usage is given in units of byte.
| Device | MIPS | Program Memory | Data Memory | Per Channel Data Memory |
| Encode | 31 | 103K | 50K | 30K |
| Decode | 13 | 71K | 18K | 30K |
AAC-LD C64x+ / C66 / C674x
CPU Utilization & Memory RequirementsAll Memory usage is given in units of byte.
| Device | MIPS | Program Memory | Data Memory | Per Channel Data Memory |
| Encode | 29.5 | 98K | 50K | 30K |
| Decode | 12.4 | 67.5K | 18K | 30K |
DESCRIPTION
Low Delay Advanced Audio Coding (AAC-LD) is the high-quality low-delay audio coding standard within MPEG-4. The MPEG-4 Low Delay Audio Coder is designed to combine the advantages of perceptual audio coding with the low delay necessary for two-way communication. It has achieved wide acceptance in high end video conference terminals and is used in professional broadcasting applications. The codec is closely derived from MPEG-4 Advanced Audio Coding (AAC-LC). It features an algorithmic delay of only 20 ms while offering good compression ratios and high sound quality audio quality for all kinds of audio signals including speech, music and atmospheric sounds. This way, AAC-LD bridges the gap between speech coding schemes and high quality audio coding schemes. Unlike common speech coders, the achieved coding quality scales up with bitrate, and transparent quality can be achieved.
MPEG4 AAC Encoder Audio Software
Sampling Rate: The MPEG4 AAC Encoder supports sampling frequency from 8kHz to 96 kHz as specified by the standard.
Bit Rate: The MPEG4 AAC Encoder supports all bit rates specified by the standard.
FEATURES
- The given Encoder implements AAC Low complexity (LC) profile of the standard up to 1channels (mono) and 2 channels (stereo).
- The given encoder supports; ADIF (Audio Data Interchange Format), ADTS (Audio Data Transport Stream) and RAW output formats.
- It supports sampling frequency from 8kHz to 96 kHz as specified by the standard.
- It supports all necessary tools and features, so that the given algorithm is standard compliant.
- Compliant with XDAIS specification.
- Rev2/Rev3 specific optimization.
MPEG4 AAC Decoder Audio Software
Sampling Rate: 8 – 96 kHz
FEATURES
- MPEG4 AAC Low complexity (LC) Object type of the standard for up to 2 channels (stereo).
- MPEG4 AAC Long Term Prediction (LTP) Object type of the standard for up to 2 channels (stereo).
- Supports both ADIF (Audio Data Interchange Format) and ADTS (Audio Data Transport Stream) encoded data.
- Compliant with XDAIS specification.
- Data memory can be allocated at run-time.
Bit Rate: The MPEG4 AAC Decoder supports all bit rates specified by the standard.
Availability
| Platforms |
| Texas Instruments – TI TMS320C5000 C55x |
ADT MPEG4 AAC Encode is available on the above Platforms: Other configurations are available upon request.
SPECIFICATIONS
DESCRIPTION
The Adaptive Digital Technologies MPEG4-AAC (LC/LTP) Decoder decodes MPEG4 AAC bit streams. The AAC decoder supports MPEG4 audio decoding for ADIF (audio data interchange format) and ADTS (audio data transport stream) encoded data.
MPEG Layer-3 Decoder Audio Software
Sampling Rate: 8 – 48 kHz
FEATURES
- Supports VBR (variable bit rate) and CBR (constant bit rate) modes.
- Supports all stereo modes (stereo/joint stereo/dual channel/single channel).
- Fully compliant with ISO/IEC 11172-3, ISO/IEC 13818-3.2 (MPEG2 audio) and MPEG2.5 standards.
- Compliant with XDAIS specification.
- Supports all mode extensions for joint stereo.
Sampling Rate: The MPEG4 MP3 Decoder supports sampling frequency from 8kHz to 48 kHz as specified by the standard.
Bit Rate: The MPEG4 MP3 Decoder supports all bit rates as specified by the standard.
Availability
| Platforms |
| Texas Instruments – TI TMS320C5000 C55x |
ADT MP3 Decode is available on the above Platforms: Other configurations are available upon request.
SPECIFICATIONS
DESCRIPTION
The Adaptive Digital Technologies MP3 Decoder is fully compliant with ISO/IEC 11172-3 (MPEG1 layer 3), ISO/IEC 13818-3.2 (MPEG2 audio) and MPEG2.5 standards as specified by ISO/IEC-11172-4 standard. The MP3 decoder converts audio data in compressed format to uncompressed format. It supports sampling frequencies from 8 kHz to 48 kHz and all bit-rates specified by the standard.
Windows Media Audio Decoder Software
FEATURES
- Supports DRM (Digital Right Management).
- Supports 2 channels in stereo mode (stereo/mono).
- Support 8-48 kHz output sampling rates and 5-384 kbps input bit rate
- Bit-Compliant as per the Microsoft test acceptance criterion.
- Compliant with XDAIS specification.
AVAILABILITY
Availability
| Platforms |
| Texas Instruments – TI TMS320C5000 C55x / C54x |
ADT WMA Decode is available on the above Platforms: Other configurations are available upon request.
SPECIFICATIONS
DESCRIPTION
The Adaptive Digital Technologies WMA Decoder is Class 4 implementation of WMA decoder, fully support all versions of WMA namely V2, V7, V8, V9, V9beta, V9NC and VBR(Variable Bit Rate) . The WMA decoder converts audio data in compressed format to uncompressed format(16bit raw PCM). It supports sampling frequencies from 8 kHz to 48 kHz and all bit-rates specified by the standard.
SPECIFICATIONS
Sampling Rate: The WMA Decoder supports sampling frequency from 8kHz to 48 kHz as specified
by the standard.
Bit Rate: The WMA Decoder supports all bit rates as specified by the standard.
Function APIs
API function call summary
ENCODE_create(…) creates an instance of the encode algorithm and returns a handle to the object
ENCODE_encode(…) processes the encoder
ALG_delete(…) delete encode algorithm instance
DECODE_create(…) creates an instance of the DECODE algorithm and returns a handle to the object.
DECODE_decode(…) processes the calls to the decoder.
DECODE_delete(…) deletes the instance of the dynamically created object referenced to by the handle