itu t g 722 1 annex c a new low complexity 14 khz audio coding standard
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ITU-T G.722.1 ANNEX C A NEW LOW-COMPLEXITY 14 KHZ AUDIO CODING STANDARD. Minjie Xie, Dave Lindbergh, and Peter Chu. ICASSP 2006. G.722.1C: First ITU-T Super-wideband Audio Coding Standard. Audio bandwidth: 14 kHz Sample rate: 32 kHz Bit rate: 24, 32, and 48 kbit/s

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itu t g 722 1 annex c a new low complexity 14 khz audio coding standard

ITU-T G.722.1 ANNEX CA NEW LOW-COMPLEXITY14 KHZ AUDIO CODING STANDARD

Minjie Xie, Dave Lindbergh, and Peter Chu

ICASSP 2006

g 722 1c first itu t super wideband audio coding standard
G.722.1C: First ITU-T Super-wideband Audio Coding Standard
  • Audio bandwidth: 14 kHz
  • Sample rate: 32 kHz
  • Bit rate: 24, 32, and 48 kbit/s
  • Algorithm: Transform coding (Siren14TM)
  • Frame size: 20 ms
  • Algorithmic delay: 40 ms
  • Complexity: <11 WMOPS (encoder+decoder)
  • Very high audio quality
  • Suitable for video and teleconferencing and Internet streaming
  • Available on royalty-free licensing terms

ICASSP 2006

overview of main g 722 1 mode
Overview of Main G.722.1 Mode
  • Wideband coding standard approved by ITU-T in 1998
  • Provides 50-7000 Hz audio bandwidth at 24 and 32 kbit/s
  • Based on transform coding, using a Modulated Lapped Transform (MLT)
  • Operates on frames of 20 ms corresponding to 320 samples at a 16 kHz sampling rate
  • A Look-ahead of 20 ms due to 50% overlap between frames
  • Total algorithmic delay of 40 ms
  • Very low computational complexity (about 5.3 WMOPS)

ICASSP 2006

g 722 1c extension mode of g 722 1
G.722.1C : Extension Mode of G.722.1
  • Audio signal sampled at 32 kHz
  • Double the audio bandwidth from 7 kHz to 14 kHz
  • Same algorithmic steps as the main mode of G.722.1
  • Same frame size as G.722.1 – 20 ms
  • Total algorithmic delay of 40 ms

ICASSP 2006

encoder of g 722 1 annex c
Encoder of G.722.1 Annex C
  • Double the MLT transform length from 320 to 640 samples
  • Double the number of frequency regions from 14 to 28
  • Double the Huffman coding tables for encoding quantized region power indices
  • Double the threshold for adjusting the number of available bits from 320 to 640

ICASSP 2006

decoder of g 722 1 annex c
Decoder of G.722.1 Annex C
  • Double the number of frequency regions from 14 to 28
  • Double the threshold for adjusting the number of available bits from 320 to 640
  • Extend the centroid table for reconstruction of MLT coefficients
  • Double the IMLT transform length from 320 to 640 samples

ICASSP 2006

computational complexity and memory requirements of g 722 1c
Computational Complexity andMemory Requirements of G.722.1C

Computational complexity

Memory requirements

ICASSP 2006

algorithmic delay of g 722 1c versus the 3gpp audio codecs
Algorithmic Delay of G.722.1C versus the 3GPP Audio Codecs

Note 1: Without bit-reservoir (see 3GPP TR 26.936 V6.1.0)

Note 2: ISF = 25.6 kHz (see 3GPP TR 26.936 V6.1.0)

ICASSP 2006

itu t subjective characterization tests
ITU-T Subjective Characterization Tests
  • Subjective tests performed by France Telecom according to a test plan designed by ITU-T SG12 SQEG
  • Characterization test Phase 1 : Speech

- ACR for clean speech and DCR for noisy speech

  • Characterization test Phase 2 : Music and mixed content

- MUSHRA method

  • Reference codec : MPEG-4 AAC-LD PCEnc/DecPro
  • Additional reference Codecs : 3GPP eAAC+ and

AMR-WB+

  • Requirements : Not worse than the reference codec for a 99% confidence interval

ICASSP 2006

conclusion
Conclusion
  • G.722.1C met all performance requirements
  • Phase 1 (clean and noisy speech)

- 24 kbit/s: Better than AAC-LD and Not Worse than eAAC+

- 32 kbit/s: Better than AAC-LD, Not Worse than eAAC+, and Not Worse than AMR-WB+ in most of tests

- 48 kbit/s: Not Worse than AAC-LD at 48 and 64 kbit/s

  • Phase 2 (music and mixed content)

- 24 kbit/s: Better than AAC-LD

- 32 kbit/s: Better than AAC-LD

- 48 kbit/s: Better than AAC-LD at 48 and 64 kbit/s

  • Executables, audio samples, and more information available at : http://www.polycom.com/Siren14

ICASSP 2006

acknowledgment
Acknowledgment

The authors would like to acknowledge Claude Lamblin, ITU-T Q.10/SG16 Rapporteur, and Catherine Quinquis, ITU-T Q.7/SG12 Rapporteur, for their great work guiding this project to a completion. In addition, the authors would like to thank the speech quality experts and staff who performed the subjective characterization tests at France Telecom.

ICASSP 2006

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