Add mute state field to AudioFrame.

webrtc/modules/audio_coding/test/delay_test.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 191 matching lines @@
         fflush (stdout);
       }
 
       in_file_a_.Read10MsData(audio_frame);
       ASSERT_GE(acm_a_->Add10MsData(audio_frame), 0);
       bool muted;
       ASSERT_EQ(0,
                 acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted));
       RTC_DCHECK(!muted);
       out_file_b_.Write10MsData(
-          audio_frame.data_,
+          audio_frame.data(),
           audio_frame.samples_per_channel_ * audio_frame.num_channels_);
       received_ts = channel_a2b_->LastInTimestamp();
       rtc::Optional<uint32_t> playout_timestamp = acm_b_->PlayoutTimestamp();
       ASSERT_TRUE(playout_timestamp);
       inst_delay_sec = static_cast<uint32_t>(received_ts - *playout_timestamp) /
                        static_cast<double>(encoding_sample_rate_hz_);
 
       if (num_frames > 10)
         average_delay = 0.95 * average_delay + 0.05 * inst_delay_sec;
 
@@ skipping 36 matching lines @@
   test_setting.codec.num_channels = FLAGS_num_channels;
   test_setting.acm.dtx = FLAGS_dtx;
   test_setting.acm.fec = FLAGS_fec;
   test_setting.packet_loss = FLAGS_packet_loss;
 
   webrtc::DelayTest delay_test;
   delay_test.Initialize();
   delay_test.Perform(&test_setting, 1, 240, "delay_test");
   return 0;
 }