Added logging for audio send/receive stream configs.

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 2cc296dd36e9de80417f817d38e66afae6ccee18..81a206ae607b53b86439f5751182665f07c50223 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -378,6 +378,7 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(
     const webrtc::AudioSendStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  event_log_->LogAudioSendStreamConfig(config);
   AudioSendStream* send_stream = new AudioSendStream(
       config, config_.audio_state, &worker_queue_, congestion_controller_.get(),
       bitrate_allocator_.get());
@@ -415,6 +416,7 @@ webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
     const webrtc::AudioReceiveStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+  event_log_->LogAudioReceiveStreamConfig(config);
   AudioReceiveStream* receive_stream =
       new AudioReceiveStream(congestion_controller_.get(), config,
                              config_.audio_state, event_log_.get());