URL: https://code.wireshark.org/review/gitweb?p=wireshark.git;a=commit;h=46370b3aea2642a140bce9a57a9318599b959b23
Submitter: Pascal Quantin (pascal.quantin@xxxxxxxxx)
Changed: branch: master
Repository: wireshark
Commits:
46370b3 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>
Actions performed:
from 7e18954 Qt: fix generation of silence samples
adds 46370b3 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(-)