[Wireshark code review <code-review-do-not-reply@xxxxxxxxxxxxx>]

URL: https://code.wireshark.org/review/gitweb?p=wireshark.git;a=commit;h=53c381e6bfd70fa0ff545c6277dd5f8d53c458b6
Submitter: Pascal Quantin (pascal.quantin@xxxxxxxxx)
Changed: branch: master-2.0
Repository: wireshark

Commits:

53c381e by Pascal Quantin (pascal.quantin@xxxxxxxxx):

    Qt: write number of decoded bytes in the RTP player temporary buffer
    
    For codecs using compression (so not G.711) the number of decoded bytes is different from payload len * sample bytes.
    This result in a truncated audio buffer and inaudible audio.
    
    Change-Id: I755c19df37820c1c56acc7bd7b67fcc104516474
    Reviewed-on: https://code.wireshark.org/review/12336
    Reviewed-by: Pascal Quantin <pascal.quantin@xxxxxxxxx>
    (cherry picked from commit 46370b3aea2642a140bce9a57a9318599b959b23)
    Reviewed-on: https://code.wireshark.org/review/12345
    

Actions performed:

    from  03aba90   Qt: fix generation of silence samples
    adds  53c381e   Qt: write number of decoded bytes in the RTP player temporary buffer


Summary of changes:
 ui/qt/rtp_audio_stream.cpp |    2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)