diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
index 4b53a524e9..b7289fc7cb 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
@@ -1000,6 +1000,9 @@ TEST_F(AcmReceiverBitExactnessOldApi, MAYBE_48kHzOutputExternalDecoder) {
   EXPECT_CALL(mock_decoder, HasDecodePlc())
       .Times(AtLeast(1))
       .WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::HasDecodePlc));
+  EXPECT_CALL(mock_decoder, PacketDuration(_, _))
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::PacketDuration));
   ExternalDecoder ed;
   ed.rtp_payload_type = 0;
   ed.external_decoder = &mock_decoder;
diff --git a/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h b/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h
index 90f132b0f7..8debcbbb1e 100644
--- a/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h
+++ b/webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h
@@ -31,6 +31,7 @@ class MockAudioDecoder : public AudioDecoder {
   MOCK_METHOD5(IncomingPacket, int(const uint8_t*, size_t, uint16_t, uint32_t,
                                    uint32_t));
   MOCK_METHOD0(ErrorCode, int());
+  MOCK_CONST_METHOD2(PacketDuration, int(const uint8_t*, size_t));
   MOCK_CONST_METHOD0(Channels, size_t());
   MOCK_CONST_METHOD0(codec_type, NetEqDecoder());
   MOCK_METHOD1(CodecSupported, bool(NetEqDecoder));
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index beadedadb4..e6f7e60dbd 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -834,6 +834,16 @@ int NetEqImpl::GetAudioInternal(size_t max_length,
   LOG(LS_VERBOSE) << "Sync buffer (" << *num_channels << " channel(s)):" <<
       " insert " << algorithm_buffer_->Size() << " samples, extract " <<
       samples_from_sync << " samples";
+  if (sync_buffer_->FutureLength() < expand_->overlap_length()) {
+    // The sync buffer should always contain |overlap_length| samples, but now
+    // too many samples have been extracted. Reinstall the |overlap_length|
+    // lookahead by moving the index.
+    const size_t missing_lookahead_samples =
+        expand_->overlap_length() - sync_buffer_->FutureLength();
+    DCHECK_GE(sync_buffer_->next_index(), missing_lookahead_samples);
+    sync_buffer_->set_next_index(sync_buffer_->next_index() -
+                                 missing_lookahead_samples);
+  }
   if (samples_from_sync != output_size_samples_) {
     LOG(LS_ERROR) << "samples_from_sync (" << samples_from_sync
                   << ") != output_size_samples_ (" << output_size_samples_
@@ -846,7 +856,7 @@ int NetEqImpl::GetAudioInternal(size_t max_length,
   *samples_per_channel = output_size_samples_;
 
   // Should always have overlap samples left in the |sync_buffer_|.
-  assert(sync_buffer_->FutureLength() >= expand_->overlap_length());
+  DCHECK_GE(sync_buffer_->FutureLength(), expand_->overlap_length());
 
   if (play_dtmf) {
     return_value = DtmfOverdub(dtmf_event, sync_buffer_->Channels(), output);
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
index 4f68c66467..3a3469241c 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -942,4 +942,67 @@ TEST_F(NetEqImplTest, FloodBufferAndGetNetworkStats) {
   NetEqNetworkStatistics stats;
   EXPECT_EQ(NetEq::kOK, neteq_->NetworkStatistics(&stats));
 }
+
+TEST_F(NetEqImplTest, DecodedPayloadTooShort) {
+  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;
+  const size_t kPayloadLengthSamples =
+      static_cast<size_t>(10 * kSampleRateHz / 1000);  // 10 ms.
+  const size_t kPayloadLengthBytes = 2 * kPayloadLengthSamples;
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  WebRtcRTPHeader rtp_header;
+  rtp_header.header.payloadType = kPayloadType;
+  rtp_header.header.sequenceNumber = 0x1234;
+  rtp_header.header.timestamp = 0x12345678;
+  rtp_header.header.ssrc = 0x87654321;
+
+  // Create a mock decoder object.
+  MockAudioDecoder mock_decoder;
+  EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
+  EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
+  EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
+      .WillRepeatedly(Return(0));
+  EXPECT_CALL(mock_decoder, PacketDuration(_, _))
+      .WillRepeatedly(Return(kPayloadLengthSamples));
+  int16_t dummy_output[kPayloadLengthSamples] = {0};
+  // The below expectation will make the mock decoder write
+  // |kPayloadLengthSamples| - 5 zeros to the output array, and mark it as
+  // speech. That is, the decoded length is 5 samples shorter than the expected.
+  EXPECT_CALL(mock_decoder,
+              Decode(_, kPayloadLengthBytes, kSampleRateHz, _, _, _))
+      .WillOnce(
+          DoAll(SetArrayArgument<4>(dummy_output,
+                                    dummy_output + kPayloadLengthSamples - 5),
+                SetArgPointee<5>(AudioDecoder::kSpeech),
+                Return(kPayloadLengthSamples - 5)));
+  EXPECT_EQ(NetEq::kOK,
+            neteq_->RegisterExternalDecoder(&mock_decoder, kDecoderPCM16B,
+                                            kPayloadType, kSampleRateHz));
+
+  // Insert one packet.
+  EXPECT_EQ(NetEq::kOK,
+            neteq_->InsertPacket(rtp_header, payload, kPayloadLengthBytes,
+                                 kReceiveTime));
+
+  EXPECT_EQ(5u, neteq_->sync_buffer_for_test()->FutureLength());
+
+  // Pull audio once.
+  const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
+  int16_t output[kMaxOutputSize];
+  size_t samples_per_channel;
+  int num_channels;
+  NetEqOutputType type;
+  EXPECT_EQ(NetEq::kOK,
+            neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
+                             &num_channels, &type));
+  ASSERT_EQ(kMaxOutputSize, samples_per_channel);
+  EXPECT_EQ(1, num_channels);
+  EXPECT_EQ(kOutputNormal, type);
+
+  EXPECT_CALL(mock_decoder, Die());
+}
 }  // namespace webrtc