As far as I observe, there are several reasons for this:
1. webrtc/modules/audio_coding/neteq4/neteq_impl.cc : 870
assert(new_codec_);
This is related to webrtc/modules/audio_coding/neteq4/decision_logic_normal.cc : 81
kUndefined can happen without new_codec_ being set
2. webrtc/modules/audio_coding/neteq4/neteq_impl.cc : 745
assert(sync_buffer_->FutureLength() >= expand_->overlap_length());
3. some other assert triggered.
The above happens not very often and goes away with no assertion.
3. (most common, this CL addresses this)
webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc : 201
payload_data_length = payload_length - rtp_header->header.paddingLength;
There are situations that
payload_length < rtp_header->header.paddingLength;
OLD ACM + NetEq3 can handle this:
a) webrtc/modules/audio_coding/main/source/acm_neteq.cc : 477
int16_t payload_length = static_cast<int16_t>(length_payload);
payload_length becomes negative in this situation
b) webrtc/modules/audio_coding/neteq/recin.c
WebRtcNetEQ_RecInInternal() handles negative payload length
I do not want to touch VoE, so I tried to let ACM2 and NetEq4 handle negative payload length.
This CL does not follow the exact way of OLD ACM + NetEq3. I stopped negative payload length at ACM and did not allow it go to NetEq4.
To try this, apply my uploaded patch : https://webrtc-codereview.appspot.com/2281004/
Let me know if you see better solutions.
R=henrik.lundin@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2292005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4860 4adac7df-926f-26a2-2b94-8c16560cd09d
663da0a8fc