Hello PJSIP List,
I have made a simple test to validate the G.722 shift fix from
My device produces a perfect sine at 440 Hz and 0dB (float 1.0).
With a G.711a encoder, I get the exact expected output.
But with the G.722 encoder, I get a sine of -6dB (float 0.5) initially, and only after a varying delay - I have seen 0.26 sec upto 9 seconds - it suddenly jumps up to 0dB (float 1.0). Screenshot from Audacity is attached. I have used ffmpeg version 2.8.15-0ubuntu0.16.04.1 to decode the stream:
ffmpeg -f g722 -i agcb_0dB_sinus_g722dc.raw -acodec pcm_s16le -ar 16000 agcb_0dB_sinus_g722dc.wav
I can provide the Wireshark pcap / RTP raw / wav if needed. To me, this behavior is quite undesirable because it may produce a sudden jump in volume on the receiving end. Can this be the expected behavior, or am I looking at a defect?
Thanks a lot for your consideration!
Martin
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