| Index: webrtc/call/call.cc
|
| diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
|
| index 51e0cceac98d84b8eb87774232b54075087d5228..fa2d1e7e782d977d65de8a64f5c49a56a31683c5 100644
|
| --- a/webrtc/call/call.cc
|
| +++ b/webrtc/call/call.cc
|
| @@ -710,7 +710,9 @@ void Call::OnNetworkChanged(uint32_t target_bitrate_bps, uint8_t fraction_loss,
|
| bitrate_allocator_->OnNetworkChanged(target_bitrate_bps, fraction_loss,
|
| rtt_ms);
|
|
|
| - {
|
| + // Ignore updates where the bitrate is zero because the aggregate network
|
| + // state is down.
|
| + if (target_bitrate_bps > 0) {
|
| rtc::CritScope lock(&bitrate_crit_);
|
| // We only update these stats if we have send streams, and assume that
|
| // OnNetworkChanged is called roughly with a fixed frequency.
|
|
|