Add mute state field to AudioFrame.

webrtc/modules/audio_coding/neteq/neteq_stereo_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 147 matching lines @@
         encoded_multi_channel_);
     if (frame_size_samples_ * 2 * num_channels_ != multi_payload_size_bytes_) {
       return -1;
     }
     rtp_generator_.GetRtpHeader(kPayloadTypeMulti, frame_size_samples_,
                                 &rtp_header_);
     return next_send_time;
   }
 
   virtual void VerifyOutput(size_t num_samples) {
+    const int16_t* output_data = output_.data();
+    const int16_t* output_multi_channel_data = output_multi_channel_.data();
     for (size_t i = 0; i < num_samples; ++i) {
       for (size_t j = 0; j < num_channels_; ++j) {
-        ASSERT_EQ(output_.data_[i],
-                  output_multi_channel_.data_[i * num_channels_ + j])
+        ASSERT_EQ(output_data[i],
+                  output_multi_channel_data[i * num_channels_ + j])
             << "Diff in sample " << i << ", channel " << j << ".";
       }
     }
   }
 
   virtual int GetArrivalTime(int send_time) {
     int arrival_time = last_arrival_time_ + (send_time - last_send_time_);
     last_send_time_ = send_time;
     last_arrival_time_ = arrival_time;
     return arrival_time;
@@ skipping 170 matching lines @@
         frame_index_(0) {
   }
 
   virtual bool Lost() {
     return (++frame_index_) % kLossInterval == 0;
   }
 
   // TODO(hlundin): NetEq is not giving bitexact results for these cases.
   virtual void VerifyOutput(size_t num_samples) {
     for (size_t i = 0; i < num_samples; ++i) {
+      const int16_t* output_data = output_.data();
+      const int16_t* output_multi_channel_data = output_multi_channel_.data();
       auto first_channel_sample =
-          output_multi_channel_.data_[i * num_channels_];
+          output_multi_channel_data[i * num_channels_];
       for (size_t j = 0; j < num_channels_; ++j) {
         const int kErrorMargin = 200;
-        EXPECT_NEAR(output_.data_[i],
-                    output_multi_channel_.data_[i * num_channels_ + j],
+        EXPECT_NEAR(output_data[i],
+                    output_multi_channel_data[i * num_channels_ + j],
                     kErrorMargin)
             << "Diff in sample " << i << ", channel " << j << ".";
         EXPECT_EQ(first_channel_sample,
-                  output_multi_channel_.data_[i * num_channels_ + j]);
+                  output_multi_channel_data[i * num_channels_ + j]);
       }
     }
   }
 
   int frame_index_;
 };
 
 TEST_P(NetEqStereoTestLosses, RunTest) {
   RunTest(100);
 }
@@ skipping 47 matching lines @@
 
 INSTANTIATE_TEST_CASE_P(MultiChannel,
                         NetEqStereoTestDelays,
                         ::testing::ValuesIn(GetTestParameters()));
 
 INSTANTIATE_TEST_CASE_P(MultiChannel,
                         NetEqStereoTestLosses,
                         ::testing::ValuesIn(GetTestParameters()));
 
 }  // namespace webrtc