Add encode/decode time tracing to audio_coding.

webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc

 /*
  *  Copyright (c) 2013 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.
  */
 
@@ skipping 80 matching lines @@
 
   void RunTest(int num_loops) {
     // Get next input packets (mono and multi-channel).
     uint32_t next_send_time;
     uint32_t next_arrival_time;
     do {
       next_send_time = GetNewPacket();
       next_arrival_time = GetArrivalTime(next_send_time);
     } while (Lost());  // If lost, immediately read the next packet.
 
-    EXPECT_CALL(*external_decoder_,
-                Decode(_, payload_size_bytes_, 1000 * samples_per_ms_, _, _, _))
+    EXPECT_CALL(
+        *external_decoder_,
+        DecodeInternal(_, payload_size_bytes_, 1000 * samples_per_ms_, _, _))
         .Times(NumExpectedDecodeCalls(num_loops));
 
     uint32_t time_now = 0;
     for (int k = 0; k < num_loops; ++k) {
       while (time_now >= next_arrival_time) {
         InsertPacket(rtp_header_, rtc::ArrayView<const uint8_t>(
                                       encoded_, payload_size_bytes_),
                      next_arrival_time);
         // Get next input packet.
         do {
@@ skipping 336 matching lines @@
                                                         kStartSeqeunceNumber,
                                                         kStartTimestamp,
                                                         kJumpFromTimestamp,
                                                         kJumpToTimestamp));
 
   RunTest(130);  // Run 130 laps @ 10 ms each in the test loop.
   EXPECT_EQ(kRecovered, test_state_);
 }
 
 }  // namespace webrtc