Add encode/decode time tracing to audio_coding.

webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.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.
  */
 
@@ skipping 958 matching lines @@
 }
 
 // Fails Android ARM64. https://code.google.com/p/webrtc/issues/detail?id=4199
 #if defined(WEBRTC_ANDROID) && defined(__aarch64__)
 #define MAYBE_48kHzOutputExternalDecoder DISABLED_48kHzOutputExternalDecoder
 #else
 #define MAYBE_48kHzOutputExternalDecoder 48kHzOutputExternalDecoder
 #endif
 TEST_F(AcmReceiverBitExactnessOldApi,
        IF_ALL_CODECS(MAYBE_48kHzOutputExternalDecoder)) {
+  // Class intended to forward a call from a mock DecodeInternal to Decode on
+  // the real decoder's Decode. DecodeInternal for the real decoder isn't
+  // public.
+  class DecodeForwarder {
+   public:
+    DecodeForwarder(AudioDecoder* decoder) : decoder_(decoder) {}
+    int Decode(const uint8_t* encoded,
+               size_t encoded_len,
+               int sample_rate_hz,
+               int16_t* decoded,
+               AudioDecoder::SpeechType* speech_type) {
+      return decoder_->Decode(encoded, encoded_len, sample_rate_hz,
+                              decoder_->PacketDuration(encoded, encoded_len) *
+                                  decoder_->Channels() * sizeof(int16_t),
+                              decoded, speech_type);
+    }
+
+   private:
+    AudioDecoder* const decoder_;
+  };
+
   AudioDecoderPcmU decoder(1);
+  DecodeForwarder decode_forwarder(&decoder);
   MockAudioDecoder mock_decoder;
   // Set expectations on the mock decoder and also delegate the calls to the
   // real decoder.
   EXPECT_CALL(mock_decoder, IncomingPacket(_, _, _, _, _))
       .Times(AtLeast(1))
       .WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::IncomingPacket));
   EXPECT_CALL(mock_decoder, Channels())
       .Times(AtLeast(1))
       .WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::Channels));
-  EXPECT_CALL(mock_decoder, Decode(_, _, _, _, _, _))
+  EXPECT_CALL(mock_decoder, DecodeInternal(_, _, _, _, _))
       .Times(AtLeast(1))
-      .WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::Decode));
+      .WillRepeatedly(Invoke(&decode_forwarder, &DecodeForwarder::Decode));
   EXPECT_CALL(mock_decoder, HasDecodePlc())
       .Times(AtLeast(1))
       .WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::HasDecodePlc));
   EXPECT_CALL(mock_decoder, PacketDuration(_, _))
       .Times(AtLeast(1))
       .WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::PacketDuration));
   ExternalDecoder ed;
   ed.rtp_payload_type = 0;
   ed.external_decoder = &mock_decoder;
   ed.sample_rate_hz = 8000;
@@ skipping 760 matching lines @@
   Run(16000, 8000, 1000);
 }
 
 TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) {
   Run(8000, 16000, 1000);
 }
 
 #endif
 
 }  // namespace webrtc