Add encode/decode time tracing to audio_coding.

webrtc/modules/audio_coding/codecs/audio_encoder.cc

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
+
 #include "webrtc/base/checks.h"
+#include "webrtc/base/trace_event.h"
 
 namespace webrtc {
 
 AudioEncoder::EncodedInfo::EncodedInfo() = default;
 
 AudioEncoder::EncodedInfo::~EncodedInfo() = default;
 
 int AudioEncoder::RtpTimestampRateHz() const {
   return SampleRateHz();
 }
 
 AudioEncoder::EncodedInfo AudioEncoder::Encode(
     uint32_t rtp_timestamp,
     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 =
       EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded);
   RTC_CHECK_LE(info.encoded_bytes, max_encoded_bytes);
   return info;
 }
 
 bool AudioEncoder::SetFec(bool enable) {
   return !enable;
 }
 
 bool AudioEncoder::SetDtx(bool enable) {
   return !enable;
 }
 
 bool AudioEncoder::SetApplication(Application application) {
   return false;
 }
 
 void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {}
 
 void AudioEncoder::SetProjectedPacketLossRate(double fraction) {}
 
 void AudioEncoder::SetTargetBitrate(int target_bps) {}
 
 }  // namespace webrtc