Add mute state field to AudioFrame.

webrtc/modules/audio_coding/acm2/acm_send_test.cc

 /*
  *  Copyright (c) 2014 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 68 matching lines @@
   assert(codec_registered_);
   if (filter_.test(static_cast<size_t>(payload_type_))) {
     // This payload type should be filtered out. Since the payload type is the
     // same throughout the whole test run, no packet at all will be delivered.
     // We can just as well signal that the test is over by returning NULL.
     return nullptr;
   }
   // 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);
     input_frame_.timestamp_ += static_cast<uint32_t>(input_block_size_samples_);
     if (data_to_send_) {
       // Encoded packet received.
       return CreatePacket();
     }
   }
   // Test ended.
@@ skipping 45 matching lines @@
          last_payload_vec_.size());
   std::unique_ptr<Packet> packet(
       new Packet(packet_memory, allocated_bytes, clock_.TimeInMilliseconds()));
   RTC_DCHECK(packet);
   RTC_DCHECK(packet->valid_header());
   return packet;
 }
 
 }  // namespace test
 }  // namespace webrtc