audio - Android native codec packet loss, delay and bandwidth required -


i going implement dynamic codec algorithm voip, codec(s) using included in android native library, i.e.

  • amr adaptive multi-rate narrowband audio codec, known amr or amr-nb.
  • gsm gsm full-rate audio codec, known gsm-fr, gsm 06.10, gsm, or fr.
  • gsm_efr gsm enhanced full-rate audio codec, known gsm-efr, gsm 06.60, or efr.
  • pcma g.711 a-law audio codec.
  • pcmu g.711 u-law audio codec.

(ref: http://developer.android.com/reference/android/net/rtp/audiocodec.html)

i information packet loss, packet delay or bandwidth usage (per direction), making decision when changing codec during call. however, can find information g.711. others, can't find information.

is there calculation data required? mean packet loss, delay or bandwidth, can decide codec use when network status of device changes.


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