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 Requirements
All Memory usage is given in units of byte.
DeviceMIPSProgram MemoryData MemoryPer Channel Data Memory
Encode31103K50K30K
Decode1371K18K30K

AAC-LD C64x+ / C66 / C674x

CPU Utilization & Memory Requirements
All Memory usage is given in units of byte.
DeviceMIPSProgram MemoryData MemoryPer Channel Data Memory
Encode29.598K50K30K
Decode12.467.5K18K30K


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.

Availability

Platforms
Texas Instruments – TI TMS320C5000 C55x

ADT MPEG4 AAC Encode is available on the above Platforms: Other configurations are available upon request.

SPECIFICATIONS

MPEG4 AAC – C55x

MIPS (Peak) CPU Resource and Memory Resource Requirements
CYCLES INFORMATION –Profiled on TMS320C5510
ConfigurationTest File ParametersMIPS (Max)MIPS (Avg)
MPEG4_AAC_551064kbps_44.1khz_stereo71.7463.13
MPEG4_AAC_551064kbps_44.1khz_mono41.5331.36

MEMORY STATISTICS
All Memory usage is given in units of kilobytes.
ConfigurationProgram MemoryConstantsScratchInstance
MPEG4_AAC_551049.2727.576.2542.88


DESCRIPTION

The Adaptive Digital Technologies MPEG4-AAC (LC/LTP) encoder compresses audio to low bit rates while maintaining near CD quality.

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

MPEG4 AAC Decode – C55x

MIPS (Peak) CPU Resource and Memory Resource Requirements
CYCLES INFORMATION –Profiled on TMS320C5510
ConfigurationTest File ParametersMIPS (Max)MIPS (Avg)
MPEG4_AAC_LC_Low Mem64kbps_44.1kHz9.418.65
MPEG4_AAC_LC_Low MIPS64kbps_44.1kHz7.016.33
MPEG4_AAC_LTP_Low Mem64kbps_44.1kHz9.418.57
MPEG4_AAC_LTP_Low MIPS64kbps_44.1kHz7.026.22
MEMORY STATISTICS
All Memory usage is given in units of kilobytes.
ConfigurationProgram MemoryConstantsScratchInstance
Funtion Program Channel Scratch Tables
MPEG4_AAC_LC_Low Mem
15.2710.228.007.48
MPEG4_AAC_LC_Low MIPS14.6125.428.007.48
MPEG4_AAC_LTP_Low Mem21.4611.3512.0017.48
MPEG4_AAC_LTP_Low MIPS21.1925.4412.0017.48

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

MPEG3 Decode C55x

CYCLES INFORMATION –Profiled on TMS320C5510
ConfigurationTest File ParametersMIPS (Max)MIPS (Avg)
MP3_DEC_ Low MIP64kbps_320 kbps18.1916.22
MP3_DEC_ Low MEM64kbps_320 kbps20.6718.74
MEMORY STATISTICS
All Memory usage is given in units of kilobytes.
ConfigurationProgram MemoryConstantsScratchInstance
MP3_DEC_ Low MIP18.7513.709.009.49
MP3_DEC_ Low MEM14.3614.384.59.49

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

WMA Decode C55x

CYCLES INFORMATION –Profiled on TMS320C5510
Measured with number of frames = 500, all internal memory.
ConfigurationTest File ParametersMIPS (Max)MIPS (Avg)
WMA decoder (non-DRM)48kHz – 320 kbps55.1316.22
WMA decoder (non-DRM)48kHz – 192 kbps34.7223.66
WMA decoder (with DRM)48kHz – 320 kbps47.9228.50
MEMORY STATISTICS
All Memory usage is given in units of kilobytes.
ConfigurationProgram MemoryConstantsScratchInstance
WMA decoder (non-DRM)28.0333.788.726.72
WMA decoder (with DRM)51.3138.058.729.74

WMA Decode C54x

CYCLES INFORMATION –Profiled on TMS320C5510
Measured with number of frames = 500, all internal memory.
ConfigurationTest File ParametersMIPS (Max)MIPS (Avg)
WMA decoder (non-DRM)48kHz – 320 kbps78.140.0
WMA decoder (non-DRM)48kHz – 192 kbps69.535.7
WMA decoder (with DRM)48kHz – 320 kbps100.091.9
MEMORY STATISTICS
All Memory usage is given in units of kilobytes.
ConfigurationProgram MemoryConstantsScratchInstance
WMA decoder (non-DRM)158.17.320
WMA decoder (with DRM)3010.27.322

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

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