Add mute state field to AudioFrame.

webrtc/modules/audio_coding/acm2/acm_receiver_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 85 matching lines @@
       auto current_codec = acm_->SendCodec();
       ASSERT_TRUE(current_codec);
       if (!CodecsEqual(codec, *current_codec))
         ASSERT_EQ(0, acm_->RegisterSendCodec(codec));
     }
     AudioFrame frame;
     // Frame setup according to the codec.
     frame.sample_rate_hz_ = codec.plfreq;
     frame.samples_per_channel_ = codec.plfreq / 100;  // 10 ms.
     frame.num_channels_ = codec.channels;
-    memset(frame.data_, 0, frame.samples_per_channel_ * frame.num_channels_ *
-           sizeof(int16_t));
+    frame.Mute();
     packet_sent_ = false;
     last_packet_send_timestamp_ = timestamp_;
     while (!packet_sent_) {
       frame.timestamp_ = timestamp_;
       timestamp_ += frame.samples_per_channel_;
       ASSERT_GE(acm_->Add10MsData(frame), 0);
     }
   }
 
   template <size_t N>
@@ skipping 382 matching lines @@
               receiver_->last_packet_sample_rate_hz());
     EXPECT_EQ(0, receiver_->LastAudioCodec(&codec));
     EXPECT_TRUE(CodecsEqual(c.inst, codec));
   }
 }
 #endif
 
 }  // namespace acm2
 
 }  // namespace webrtc