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webrtc / src.git / 1e91693fe293cdf708ea6d3c20a289fd122799bc / . / src / modules / audio_coding / codecs / iSAC / fix / source
tree: 06473c0a8d1b849b96cb05f770f2343db228e4b1 [path history] [tgz]
  1. Android.mk
  2. arith_routines.c
  3. arith_routines_hist.c
  4. arith_routines_logist.c
  5. arith_routins.h
  6. bandwidth_estimator.c
  7. bandwidth_estimator.h
  8. codec.h
  9. decode.c
  10. decode_bwe.c
  11. decode_plc.c
  12. encode.c
  13. entropy_coding.c
  14. entropy_coding.h
  15. fft.c
  16. fft.h
  17. filterbank_tables.c
  18. filterbank_tables.h
  19. filterbanks.c
  20. filters.c
  21. filters_neon.c
  22. initialize.c
  23. isacfix.c
  24. isacfix.gypi
  25. lattice.c
  26. lpc_masking_model.c
  27. lpc_masking_model.h
  28. lpc_tables.c
  29. lpc_tables.h
  30. pitch_estimator.c
  31. pitch_estimator.h
  32. pitch_filter.c
  33. pitch_gain_tables.c
  34. pitch_gain_tables.h
  35. pitch_lag_tables.c
  36. pitch_lag_tables.h
  37. settings.h
  38. spectrum_ar_model_tables.c
  39. spectrum_ar_model_tables.h
  40. structs.h
  41. transform.c