Add mute state field to AudioFrame.

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 182 matching lines @@
     bool muted;
     EXPECT_EQ(NetEq::kOK, neteq_internal_->GetAudio(&output_internal_, &muted));
     ASSERT_FALSE(muted);
     EXPECT_EQ(1u, output_internal_.num_channels_);
     EXPECT_EQ(static_cast<size_t>(kOutputLengthMs * sample_rate_hz_ / 1000),
               output_internal_.samples_per_channel_);
 
     // Get audio from external decoder instance.
     GetOutputAudio(&output_);
 
+    const int16_t* output_data = output_.data();
+    const int16_t* output_internal_data = output_internal_.data();
     for (size_t i = 0; i < output_.samples_per_channel_; ++i) {
-      ASSERT_EQ(output_.data_[i], output_internal_.data_[i])
+      ASSERT_EQ(output_data[i], output_internal_data[i])
           << "Diff in sample " << i << ".";
     }
   }
 
   void InsertPacket(RTPHeader rtp_header,
                     rtc::ArrayView<const uint8_t> payload,
                     uint32_t receive_timestamp) override {
     // Insert packet in internal decoder.
     ASSERT_EQ(NetEq::kOK, neteq_internal_->InsertPacket(rtp_header, payload,
                                                         receive_timestamp));
@@ skipping 76 matching lines @@
     AudioFrame output;
     GetOutputAudio(&output);
     UpdateState(output.speech_type_);
 
     if (test_state_ == kExpandPhase || test_state_ == kFadedExpandPhase) {
       // Don't verify the output in this phase of the test.
       return;
     }
 
     ASSERT_EQ(1u, output.num_channels_);
+    const int16_t* output_data = output.data();
     for (size_t i = 0; i < output.samples_per_channel_; ++i) {
-      if (output.data_[i] != 0)
+      if (output_data[i] != 0)
         return;
     }
     EXPECT_TRUE(false)
         << "Expected at least one non-zero sample in each output block.";
   }
 
   int NumExpectedDecodeCalls(int num_loops) override {
     // Some packets at the end of the stream won't be decoded. When the jump in
     // timestamp happens, NetEq will do Expand during one GetAudio call. In the
     // next call it will decode the packet after the jump, but the net result is
@@ skipping 132 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