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( |