diff options
| author | Pascal Quantin <pascal.quantin@gmail.com> | 2015-12-01 19:23:32 +0100 |
|---|---|---|
| committer | Pascal Quantin <pascal.quantin@gmail.com> | 2015-12-01 21:24:42 +0000 |
| commit | 46370b3aea2642a140bce9a57a9318599b959b23 (patch) | |
| tree | 2a5006bc01c9a49b0b490194591a1bfa57e2233c /ui/qt/rtp_audio_stream.cpp | |
| parent | 7e18954a276f93c78e4ae7129ad97d73ec6d91aa (diff) | |
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@gmail.com>
Diffstat (limited to 'ui/qt/rtp_audio_stream.cpp')
| -rw-r--r-- | ui/qt/rtp_audio_stream.cpp | 2 |
1 files changed, 1 insertions, 1 deletions
diff --git a/ui/qt/rtp_audio_stream.cpp b/ui/qt/rtp_audio_stream.cpp index 99381e6509..a8f4c02011 100644 --- a/ui/qt/rtp_audio_stream.cpp +++ b/ui/qt/rtp_audio_stream.cpp @@ -319,7 +319,7 @@ void RtpAudioStream::decode() // Write samples to our file. write_buff = (char *) decode_buff; - write_bytes = rtp_packet->info->info_payload_len * sample_bytes_; + write_bytes = decoded_bytes; if (audio_out_rate_ != sample_rate) { // Resample the audio to match our previous output rate. |