Handle padded audio packets correctly

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 1266 matching lines @@
   CreateInstance();
   // Let the dummy target delay be 17 packets.
   constexpr int kTargetLevelPacketsQ8 = 17 << 8;
   EXPECT_CALL(*mock_delay_manager_, TargetLevel())
       .WillOnce(Return(kTargetLevelPacketsQ8));
   // Default packet size before any packet has been decoded is 30 ms, so we are
   // expecting 17 * 30 = 510 ms target delay.
   EXPECT_EQ(17 * 30, neteq_->TargetDelayMs());
 }
 
+TEST_F(NetEqImplTest, InsertEmptyPacket) {
+  UseNoMocks();
+  use_mock_delay_manager_ = true;
+  CreateInstance();
+
+  RTPHeader rtp_header;
+  rtp_header.payloadType = 17;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  EXPECT_CALL(*mock_delay_manager_, RegisterEmptyPacket());
+  neteq_->InsertEmptyPacket(rtp_header);
+}
+
 class Decoder120ms : public AudioDecoder {
  public:
   Decoder120ms(int sample_rate_hz, SpeechType speech_type)
       : sample_rate_hz_(sample_rate_hz),
         next_value_(1),
         speech_type_(speech_type) {}
 
   int DecodeInternal(const uint8_t* encoded,
                      size_t encoded_len,
                      int sample_rate_hz,
@@ skipping 205 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