EVS

EVS

CODEC OVERVIEW

Enhanced Voice Services (EVS) speech coding standard was developed by 3GPP in the year 2014. The codec operates on 20ms frames of 16-bit PCM speech/audio signals sampled at 8 KHz, 16 KHz, 32 KHz, or 48 KHz and generates a compressed bit-stream having bit-rates in the range of 5.9 kbps to 128 kbps respectively. EVS has been developed primarily for VoLTE and is inter-operable with the 3GPP AMR-WB codec. At 5.9 kbps, it operates in variable bit rate and it also has a robust channel aware mode at 13.2 kbps for lossy networks. All other bit-rates operate in constant bit-rate mode. Encoder uses the ACELP core for speech input, DTX mode for inactive input, and MDCT mode for audio input. The decoder supports packet loss concealment and has an inherent jitter buffer management for handling packet loss and delay jitter.

SALIENT FEATURES
  • Based on EVS standard Release 12.1
  • Fixed-point ANSI C implementation on TI platform.
  • Floating-point ANSI C implementation on AMD/Intel platform.
  • Re-entrant implementation
  • C-callable APIs
  • Optimized C/Assembly implementation
  • Operates on 20 ms frame length
  • Operates on speech/audio signals sampled at 8 KHz, 16KHz, 32KHz, and 48 KHz
  • Supports bit-rates ranging from 5.9 kbps to 128 kbps
  • Supports Constant Bit Rate (CBR), Variable Bit Rate (VBR) and channel aware modes
  • Supports inter-operability with the 3GPP AMR-WB codec
  • Supports configuring bandwidth at int- and run-time.
  • Supports configuring the bit-rate at init- and run-time
  • Supports Forward Error Correction (FEC) for good robustness
  • Supports integrated Packet Loss Concealment (PLC) algorithm
  • Supports VAD/DTX/CNG configurable at init- and run-time
  • Optional support for xDM APIs on TI platform
TESTING FEATURES
  • Fixed-point implementation is tested for bit-exactness with standard as well as a large database of non-standard test vectors.
  • Module is fully interruptible.
  • C64x+/C66x implementations tested for any illegal memory access.
  • Tested for compliance with register preservation requirements
  • Tested for Input buffer corruption
  • Tested for I/O buffer alignment requirements
  • Tested for multi-instance implementation
  • Tested for 100% code coverage
  • Range validation for all the API parameters
  • Tested with scratch contamination at frame boundaries
  • Tested for packet loss conditions with 5% loss to 25% loss
  • C64x+ implementation validated on C6472EVM platform.
  • C66x implementation validated on C6678EVM platform.
  • AMD/Intel optimized implementation validated on Intel cores supporting SSE4 and above.
AVAILABLE PLATFORM(S)

TI C66x, TI C64x+, and AMD/Intel 64-bit cores supporting SSE4 and above.

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G.711

G.711

CODEC OVERVIEW

G.711 speech codec was standardized by ITU-T in 1972. The codec operates on each 16-bit speech signal sampled at 8 KHz and generates a compressed 8-bit sample resulting in an overall bit-rate of 64 kbps. It compresses the signal using either the mu-law or A-law algorithm. Appendix I, introduced in 1999, specifies a simple low complexity packet loss concealment algorithm at the decoder. Appendix II, introduced in 2000, defines the comfort noise payload definition for discontinuous transmission systems (DTX) at the encoder. The codec is used for benchmarking the performance of other speech codecs and is used as the default codec in VoIP applications.

SALIENT FEATURES
  • Based on ITU-T specification.
  • Optimized ASM/C implementation.
  • Re-entrant implementation.
  • C-callable APIs.
  • Operates on speech/audio signals sampled at 8 KHz.
  • Support for 64 kbps bit-rate.
  • Support for RTP payload format as specified in RFC 3551.
  • Supports DTX mode of operation as specified in Apppendix II for 10ms frame size; configurable at init-time.
  • Supports Packet Loss Concealment (PLC) algorithm as specified in Appendix I.
  • Support for PLC for frame size of the order of 1ms.
  • The implementation supports both Little-Endian and Big-Endian (on ARM and C64x platforms)
  • Optional support for xDM APIs on TI platforms.
TESTING FEATURES
  • Tested for bit-exactness with standard as well as a large database of non-standard test vectors.
  • Module is fully interruptible (Maximum interrupt latency on C64x is 6000 cycles).
  • Tested for any illegal memory access by the module (C64x and ARM)
  • Tested for compliance with register preservation requirements.
  • Tested for Input buffer corruption.
  • Tested for I/O buffer alignment requirements.
  • Tested for multi-instance implementation.
  • Tested for 100% code coverage.
  • Range validation for all the API parameters.
  • Tested with scratch contamination at frame boundaries.
  • Tested for packet loss conditions with 5% loss to 25% loss.
  • TI C55x implementation validated on Spectrum Digital C5510 DSK.
  • TI C64x+ implementation validated on Spectrum Digital C6455 DSK.
  • ARM implementation validated on OMAP3530 (Cortex-A8) and DM6446/DM6467 (ARM926EJ-S) platforms.
  • AMD/Intel optimized implementation validated on Intel cores supporting SSE4 and above.
AVAILABLE PLATFORM(S)

ARM9E, ARM11, Cortex-A8, Cortex-A9, TI C55x,  TI C64x+, and AMD/Intel 64-bit cores supporting SSE4 and above.

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G.711.1

G.711.1

CODEC OVERVIEW

G.711.1 speech codec was standardized by ITU-T in 2008. The codec operates on each 5 ms frame of 16-bit speech/audio signals sampled at 8 KHz or 16 KHz and generates a compressed bit-stream having bit-rates of 64 kbps, 80 kbps, or 96 kbps structured as 3 layers. The layered approach allows the decoder or any other component in the communication system to truncate the bit-stream by removing the higher layers. The base layer (Layer 0), at 64 kbps, is inter-operable with the traditional G.711. Layer 1 provides a lower band enhancement bit-stream at 16 kbps. Layer 2 provides a higher band enhancement bit-stream for the 16 KHz sampling frequency at 16 kbps. To improve the quality under frame erasures due to channel errors such as packet-losses, frame erasure concealment algorithms are provided at the decoder. This codec aims to achieve high-quality speech services over broadband networks, particularly for IP phone and multi-point speech conferencing, while enabling a seamless interoperability with conventional terminals and systems equipped only with G.711.

SALIENT FEATURES
  • Based on ITU-T specification.
  • Optimized ASM/C implementation.
  • Re-entrant implementation.
  • C-callable APIs.
  • Operates on speech/audio signals sampled at 16 KHz and 8 KHz.
  • Support for 64 kbps and 80 kbps bit-rate for the 8 KHz sampling frequency.
  • Support for 64 kbps, 80 kbps, and 96kbps bit-rate for the 16 KHz sampling frequency.
  • Support for RTP payload format as specified in RFC 3551.
  • Support for the integrated frame erasure concealment algorithm.
  • Support for the RTP payload format as specified in the standard.
  • Optional support for xDM APIs on TI platforms.
TESTING FEATURES
  • Tested for bit-exactness with standard as well as a large database of non-standard test vectors.
  • Module is fully interruptible.
  • Tested for compliance with register preservation requirements.
  • Tested for Input buffer corruption.
  • Tested for I/O buffer alignment requirements.
  • Tested for multi-instance implementation.
  • Tested for 100% code coverage.
  • Range validation for all the API parameters.
  • Tested with scratch contamination at frame boundaries.
  • Tested for packet loss conditions with 5% loss to 25% loss.
  • TI C55x implementation validated on Spectrum Digital C5510 DSK.
AVAILABLE PLATFORM(S)

TI C55x.

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