Implement AudioSendStream::GetStats().

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 3969bc6b517e97c2f51bcd35a64af8cc9a40654b..63ce5f72b32f6bd44ca434fa99c9b45896204cb3 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -149,7 +149,6 @@ Call::Call(const Call::Config& config)
       network_enabled_(true),
       receive_crit_(RWLockWrapper::CreateRWLock()),
       send_crit_(RWLockWrapper::CreateRWLock()) {
-  RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
   RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
                 config.bitrate_config.min_bitrate_bps);
@@ -203,7 +202,8 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(
     const webrtc::AudioSendStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
-  AudioSendStream* send_stream = new AudioSendStream(config);
+  AudioSendStream* send_stream =
+      new AudioSendStream(config, config_.voice_engine);
   {
     rtc::CritScope lock(&network_enabled_crit_);
     WriteLockScoped write_lock(*send_crit_);