Add encode/decode time tracing to audio_coding.

webrtc/modules/audio_coding/codecs/audio_encoder.cc

Index: webrtc/modules/audio_coding/codecs/audio_encoder.cc
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.cc b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
index 388b0ff61c6d2b9d02ccebccd7779a329a52bb69..e99fc30995995dea55ba1bcae4347b784e4691f3 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.cc
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
@@ -9,7 +9,9 @@
  */
 
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
+
 #include "webrtc/base/checks.h"
+#include "webrtc/base/trace_event.h"
 
 namespace webrtc {
 
@@ -26,6 +28,7 @@ AudioEncoder::EncodedInfo AudioEncoder::Encode(
     rtc::ArrayView<const int16_t> audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
+  TRACE_EVENT0("webrtc", "AudioEncoder::Encode");
   RTC_CHECK_EQ(audio.size(),
                static_cast<size_t>(NumChannels() * SampleRateHz() / 100));
   EncodedInfo info =