Add mute state field to AudioFrame.

webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc

Index: webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
index de07ae20722a7240a20af3d2e9c00aa2ddc570aa..7d652989a03a09d9997078b6162cc6a70a386286 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -215,7 +215,7 @@ class NetEqImplTest : public ::testing::Test {
         1512, 2378, 2828, 2674, 1877, 568, -986, -2446, -3482, -3864, -3516,
         -2534, -1163 });
     ASSERT_GE(kMaxOutputSize, kOutput.size());
-    EXPECT_TRUE(std::equal(kOutput.begin(), kOutput.end(), output.data_));
+    EXPECT_TRUE(std::equal(kOutput.begin(), kOutput.end(), output.data()));
   }
 
   std::unique_ptr<NetEqImpl> neteq_;
@@ -524,7 +524,7 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
   // Wrap the expected value in an rtc::Optional to compare them as such.
   EXPECT_EQ(
       rtc::Optional<uint32_t>(rtp_header.header.timestamp +
-                              output.data_[output.samples_per_channel_ - 1]),
+                              output.data()[output.samples_per_channel_ - 1]),
       neteq_->GetPlayoutTimestamp());
 
   // Check the timestamp for the last value in the sync buffer. This should
@@ -537,7 +537,7 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
   // Check that the number of samples still to play from the sync buffer add
   // up with what was already played out.
   EXPECT_EQ(
-      kPayloadLengthSamples - output.data_[output.samples_per_channel_ - 1],
+      kPayloadLengthSamples - output.data()[output.samples_per_channel_ - 1],
       sync_buffer->FutureLength());
 }