Add mute state field to AudioFrame.

webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc

 /*
  *  Copyright (c) 2012 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 197 matching lines @@
 
     // Verify first 64 samples of actual output.
     const std::vector<int16_t> kOutput({
         0, 0, 0, 0, 0, 0, 0, 0, 0, 0, -1578, -2816, -3460, -3403, -2709, -1594,
         -363, 671, 1269, 1328, 908, 202, -513, -964, -955, -431, 504, 1617,
         2602, 3164, 3101, 2364, 1073, -511, -2047, -3198, -3721, -3525, -2688,
         -1440, -99, 1015, 1663, 1744, 1319, 588, -171, -680, -747, -315, 515,
         1512, 2378, 2828, 2674, 1877, 568, -986, -2446, -3482, -3864, -3516,
         -2534, -1163 });
     ASSERT_GE(kMaxOutputSize, kOutput.size());
-    EXPECT_TRUE(std::equal(kOutput.begin(), kOutput.end(), output.data_));
+    EXPECT_TRUE(std::equal(kOutput.begin(), kOutput.end(), output.data()));
   }
 
   std::unique_ptr<NetEqImpl> neteq_;
   NetEq::Config config_;
   TickTimer* tick_timer_ = nullptr;
   MockBufferLevelFilter* mock_buffer_level_filter_ = nullptr;
   BufferLevelFilter* buffer_level_filter_ = nullptr;
   bool use_mock_buffer_level_filter_ = true;
   MockDecoderDatabase* mock_decoder_database_ = nullptr;
   DecoderDatabase* decoder_database_ = nullptr;
@@ skipping 288 matching lines @@
   // Start with a simple check that the fake decoder is behaving as expected.
   EXPECT_EQ(kPayloadLengthSamples,
             static_cast<size_t>(decoder_.next_value() - 1));
 
   // The value of the last of the output samples is the same as the number of
   // samples played from the decoded packet. Thus, this number + the RTP
   // timestamp should match the playout timestamp.
   // Wrap the expected value in an rtc::Optional to compare them as such.
   EXPECT_EQ(
       rtc::Optional<uint32_t>(rtp_header.header.timestamp +
-                              output.data_[output.samples_per_channel_ - 1]),
+                              output.data()[output.samples_per_channel_ - 1]),
       neteq_->GetPlayoutTimestamp());
 
   // Check the timestamp for the last value in the sync buffer. This should
   // be one full frame length ahead of the RTP timestamp.
   const SyncBuffer* sync_buffer = neteq_->sync_buffer_for_test();
   ASSERT_TRUE(sync_buffer != NULL);
   EXPECT_EQ(rtp_header.header.timestamp + kPayloadLengthSamples,
             sync_buffer->end_timestamp());
 
   // Check that the number of samples still to play from the sync buffer add
   // up with what was already played out.
   EXPECT_EQ(
-      kPayloadLengthSamples - output.data_[output.samples_per_channel_ - 1],
+      kPayloadLengthSamples - output.data()[output.samples_per_channel_ - 1],
       sync_buffer->FutureLength());
 }
 
 TEST_F(NetEqImplTest, ReorderedPacket) {
   UseNoMocks();
   CreateInstance();
 
   const uint8_t kPayloadType = 17;   // Just an arbitrary number.
   const uint32_t kReceiveTime = 17;  // Value doesn't matter for this test.
   const int kSampleRateHz = 8000;
@@ skipping 938 matching lines @@
   EXPECT_CALL(*mock_delay_manager_, BufferLimits(_, _))
       .Times(1)
       .WillOnce(DoAll(SetArgPointee<0>(1), SetArgPointee<1>(2)));
 
   bool muted;
   EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
   EXPECT_EQ(kAccelerate, neteq_->last_operation_for_test());
 }
 
 }// namespace webrtc