Add mute state field to AudioFrame.

webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.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 155 matching lines @@
   void SetUp() {
     acm_.reset(AudioCodingModule::Create(id_, clock_));
 
     rtp_utility_->Populate(&rtp_header_);
 
     input_frame_.sample_rate_hz_ = kSampleRateHz;
     input_frame_.num_channels_ = 1;
     input_frame_.samples_per_channel_ = kSampleRateHz * 10 / 1000;  // 10 ms.
     static_assert(kSampleRateHz * 10 / 1000 <= AudioFrame::kMaxDataSizeSamples,
                   "audio frame too small");
-    memset(input_frame_.data_,
-           0,
-           input_frame_.samples_per_channel_ * sizeof(input_frame_.data_[0]));
+    input_frame_.Mute();
 
     ASSERT_EQ(0, acm_->RegisterTransportCallback(&packet_cb_));
 
     SetUpL16Codec();
   }
 
   // Set up L16 codec.
   virtual void SetUpL16Codec() {
     audio_format_ =
         rtc::Optional<SdpAudioFormat>(SdpAudioFormat("L16", kSampleRateHz, 1));
@@ skipping 500 matching lines @@
     ASSERT_EQ(
         0,
         acm_->IncomingPacket(
             &last_payload_vec_[0], last_payload_vec_.size(), rtp_header_));
   }
 
   void InsertAudio() override {
     // TODO(kwiberg): Use std::copy here. Might be complications because AFAICS
     // this call confuses the number of samples with the number of bytes, and
     // ends up copying only half of what it should.
-    memcpy(input_frame_.data_, audio_loop_.GetNextBlock().data(),
+    memcpy(input_frame_.mutable_data(), audio_loop_.GetNextBlock().data(),
            kNumSamples10ms);
     AudioCodingModuleTestOldApi::InsertAudio();
   }
 
   // Override the verification function with no-op, since iSAC produces variable
   // payload sizes.
   void VerifyEncoding() override {}
 
   // This method is the same as AudioCodingModuleMtTestOldApi::TestDone(), but
   // here it is using the constants defined in this class (i.e., shorter test
@@ skipping 1129 matching lines @@
   Run(16000, 8000, 1000);
 }
 
 TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) {
   Run(8000, 16000, 1000);
 }
 
 #endif
 
 }  // namespace webrtc