Index: webrtc/modules/audio_coding/neteq/rtcp.cc |
diff --git a/webrtc/modules/audio_coding/neteq/rtcp.cc b/webrtc/modules/audio_coding/neteq/rtcp.cc |
index 7ef40bc814548de74cc86df77a3f4633a76c8f54..c5a7050e8f067143943ef6783252cdb3a566b872 100644 |
--- a/webrtc/modules/audio_coding/neteq/rtcp.cc |
+++ b/webrtc/modules/audio_coding/neteq/rtcp.cc |
@@ -47,7 +47,7 @@ void Rtcp::Update(const RTPHeader& rtp_header, uint32_t receive_timestamp) { |
if (received_packets_ > 1) { |
int32_t ts_diff = receive_timestamp - (rtp_header.timestamp - transit_); |
ts_diff = WEBRTC_SPL_ABS_W32(ts_diff); |
- int32_t jitter_diff = (ts_diff << 4) - jitter_; |
+ uint64_t jitter_diff = (uint64_t{ts_diff} << 4) - jitter_; |
ossu
2016/10/31 14:11:01
Uh... I spoke too soon. Clearly this could be nega
|
// Calculate 15 * jitter_ / 16 + jitter_diff / 16 (with proper rounding). |
jitter_ = jitter_ + ((jitter_diff + 8) >> 4); |
} |