Add mute state field to AudioFrame.

webrtc/modules/audio_coding/acm2/acm_send_test.cc

Index: webrtc/modules/audio_coding/acm2/acm_send_test.cc
diff --git a/webrtc/modules/audio_coding/acm2/acm_send_test.cc b/webrtc/modules/audio_coding/acm2/acm_send_test.cc
index 5e68afc4d117ab580549d7e2547dcbae34ffbbcd..4ff9bcfcc61c37ab9a965001fa50baea764878ed 100644
--- a/webrtc/modules/audio_coding/acm2/acm_send_test.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_send_test.cc
@@ -86,13 +86,13 @@ std::unique_ptr<Packet> AcmSendTestOldApi::NextPacket() {
   // Insert audio and process until one packet is produced.
   while (clock_.TimeInMilliseconds() < test_duration_ms_) {
     clock_.AdvanceTimeMilliseconds(kBlockSizeMs);
-    RTC_CHECK(
-        audio_source_->Read(input_block_size_samples_, input_frame_.data_));
+    RTC_CHECK(audio_source_->Read(input_block_size_samples_,
+                                  input_frame_.mutable_data()));
     if (input_frame_.num_channels_ > 1) {
-      InputAudioFile::DuplicateInterleaved(input_frame_.data_,
+      InputAudioFile::DuplicateInterleaved(input_frame_.data(),
                                            input_block_size_samples_,
                                            input_frame_.num_channels_,
-                                           input_frame_.data_);
+                                           input_frame_.mutable_data());
     }
     data_to_send_ = false;
     RTC_CHECK_GE(acm_->Add10MsData(input_frame_), 0);