Added logging for audio send/receive stream configs.
BUG=webrtc:4741,webrtc:6399
Review-Url: https://codereview.webrtc.org/2353543003
Cr-Commit-Position: refs/heads/master@{#14585}
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 5d6bbab..9515ac1 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -370,6 +370,7 @@
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(), event_log_);
@@ -407,6 +408,7 @@
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_);
{
