| Index: webrtc/call/call.cc
|
| diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
|
| index 6475e00abf204e3327cd2ce6503537951c97c6e8..26d97aad52100b7443f860e35875f2921b67026a 100644
|
| --- a/webrtc/call/call.cc
|
| +++ b/webrtc/call/call.cc
|
| @@ -637,9 +637,9 @@ void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
|
|
|
| send_stream->Stop();
|
|
|
| + const uint32_t ssrc = send_stream->GetConfig().rtp.ssrc;
|
| webrtc::internal::AudioSendStream* audio_send_stream =
|
| static_cast<webrtc::internal::AudioSendStream*>(send_stream);
|
| - const uint32_t ssrc = audio_send_stream->config().rtp.ssrc;
|
| suspended_audio_send_ssrcs_[ssrc] = audio_send_stream->GetRtpState();
|
| {
|
| WriteLockScoped write_lock(*send_crit_);
|
| @@ -656,7 +656,7 @@ void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
|
| }
|
| UpdateAggregateNetworkState();
|
| sent_rtp_audio_timer_ms_.Extend(audio_send_stream->GetActiveLifetime());
|
| - delete audio_send_stream;
|
| + delete send_stream;
|
| }
|
|
|
| webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
|
|
|