diff --git a/webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h b/webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h
index f89f887f94..f3bcc711f3 100644
--- a/webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h
+++ b/webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h
@@ -124,7 +124,7 @@ class AudioDecoder {
   // applicable (e.g., for RED and DTMF/AVT types).
   static AudioDecoder* CreateAudioDecoder(NetEqDecoder codec_type);
 
-  size_t channels() { return channels_; }
+  size_t channels() const { return channels_; }
 
  protected:
   static SpeechType ConvertSpeechType(int16_t type);
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_impl.cc b/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
index cf0a1dbafd..4ed3976434 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
@@ -465,6 +465,7 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
     memcpy(&main_header, &packet->header, sizeof(main_header));
   }
 
+  bool update_sample_rate_and_channels = false;
   // Reinitialize NetEq if it's needed (changed SSRC or first call).
   if ((main_header.ssrc != ssrc_) || first_packet_) {
     rtcp_.Init(main_header.sequenceNumber);
@@ -489,6 +490,9 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
 
     // Reset timestamp scaling.
     timestamp_scaler_->Reset();
+
+    // Triger an update of sampling rate and the number of channels.
+    update_sample_rate_and_channels = true;
   }
 
   // Update RTCP statistics, only for regular packets.
@@ -598,6 +602,7 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
   if (ret == PacketBuffer::kFlushed) {
     // Reset DSP timestamp etc. if packet buffer flushed.
     new_codec_ = true;
+    update_sample_rate_and_channels = true;
     LOG_F(LS_WARNING) << "Packet buffer flushed";
   } else if (ret == PacketBuffer::kOversizePacket) {
     LOG_F(LS_WARNING) << "Packet larger than packet buffer";
@@ -615,6 +620,26 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
     }
   }
 
+  if (update_sample_rate_and_channels && !packet_buffer_->Empty()) {
+    // We do not use |current_rtp_payload_type_| to |set payload_type|, but
+    // get the next RTP header from |packet_buffer_| to obtain the payload type.
+    // The reason for it is the following corner case. If NetEq receives a
+    // CNG packet with a sample rate different than the current CNG then it
+    // flushes its buffer, assuming send codec must have been changed. However,
+    // payload type of the hypothetically new send codec is not known.
+    const RTPHeader* rtp_header = packet_buffer_->NextRtpHeader();
+    assert(rtp_header);
+    int payload_type = rtp_header->payloadType;
+    AudioDecoder* decoder = decoder_database_->GetDecoder(payload_type);
+    assert(decoder);  // Payloads are already checked to be valid.
+    const DecoderDatabase::DecoderInfo* decoder_info =
+        decoder_database_->GetDecoderInfo(payload_type);
+    assert(decoder_info);
+    if (decoder_info->fs_hz != fs_hz_ ||
+        decoder->channels() != algorithm_buffer_->Channels())
+      SetSampleRateAndChannels(decoder_info->fs_hz, decoder->channels());
+  }
+
   // TODO(hlundin): Move this code to DelayManager class.
   const DecoderDatabase::DecoderInfo* dec_info =
           decoder_database_->GetDecoderInfo(main_header.payloadType);
@@ -1110,7 +1135,14 @@ int NetEqImpl::Decode(PacketList* packet_list, Operations* operation,
           PacketBuffer::DeleteAllPackets(packet_list);
           return kDecoderNotFound;
         }
-        SetSampleRateAndChannels(decoder_info->fs_hz, decoder->channels());
+        // We should have correct sampling rate and number of channels. They
+        // are set when packets are inserted.
+        if (decoder_info->fs_hz != fs_hz_ ||
+            decoder->channels() != algorithm_buffer_->Channels()) {
+          LOG_F(LS_ERROR) << "Sampling rate or number of channels mismatch.";
+          assert(false);
+          SetSampleRateAndChannels(decoder_info->fs_hz, decoder->channels());
+        }
         sync_buffer_->set_end_timestamp(timestamp_);
         playout_timestamp_ = timestamp_;
       }
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_impl_unittest.cc b/webrtc/modules/audio_coding/neteq4/neteq_impl_unittest.cc
index 47fa18e200..7a82053918 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_impl_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq4/neteq_impl_unittest.cc
@@ -159,7 +159,7 @@ TEST_F(NetEqImplTest, InsertPacket) {
   EXPECT_CALL(*decoder_database_, IsDtmf(kPayloadType))
       .WillRepeatedly(Return(false));  // This is not DTMF.
   EXPECT_CALL(*decoder_database_, GetDecoder(kPayloadType))
-      .Times(2)
+      .Times(3)
       .WillRepeatedly(Return(&mock_decoder));
   EXPECT_CALL(*decoder_database_, IsComfortNoise(kPayloadType))
       .WillRepeatedly(Return(false));  // This is not CNG.
@@ -183,6 +183,9 @@ TEST_F(NetEqImplTest, InsertPacket) {
   // index) is a pointer, and the variable pointed to is set to kPayloadType.
   // Also invoke the function DeletePacketsAndReturnOk to properly delete all
   // packets in the list (to avoid memory leaks in the test).
+  EXPECT_CALL(*packet_buffer_, NextRtpHeader())
+      .Times(1)
+      .WillOnce(Return(&rtp_header.header));
 
   // Expectations for DTMF buffer.
   EXPECT_CALL(*dtmf_buffer_, Flush())
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
index cefa3bb943..bc2a9de17b 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
@@ -21,6 +21,7 @@
 #include <string>
 #include <vector>
 
+#include "gflags/gflags.h"
 #include "gtest/gtest.h"
 #include "webrtc/modules/audio_coding/neteq4/test/NETEQTEST_RTPpacket.h"
 #include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
@@ -28,6 +29,8 @@
 #include "webrtc/test/testsupport/gtest_disable.h"
 #include "webrtc/typedefs.h"
 
+DEFINE_bool(gen_ref, false, "Generate reference files.");
+
 namespace webrtc {
 
 static bool IsAllZero(const int16_t* buf, int buf_length) {
@@ -313,7 +316,7 @@ void NetEqDecodingTest::DecodeAndCompare(const std::string &rtp_file,
 
   std::string ref_out_file = "";
   if (ref_file.empty()) {
-    ref_out_file = webrtc::test::OutputPath() + "neteq_out.pcm";
+    ref_out_file = webrtc::test::OutputPath() + "neteq_universal_ref.pcm";
   }
   RefFiles ref_files(ref_file, ref_out_file);
 
@@ -508,12 +511,17 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(MAYBE_TestBitExactness)) {
   // For Visual Studio 2012 and later, we will have to use the generic reference
   // file, rather than the windows-specific one.
   const std::string kInputRefFile = webrtc::test::ProjectRootPath() +
-      "resources/audio_coding/neteq_universal_ref.pcm";
+      "resources/audio_coding/neteq4_universal_ref.pcm";
 #else
   const std::string kInputRefFile =
-      webrtc::test::ResourcePath("audio_coding/neteq_universal_ref", "pcm");
+      webrtc::test::ResourcePath("audio_coding/neteq4_universal_ref", "pcm");
 #endif
-  DecodeAndCompare(kInputRtpFile, kInputRefFile);
+
+  if (FLAGS_gen_ref) {
+    DecodeAndCompare(kInputRtpFile, "");
+  } else {
+    DecodeAndCompare(kInputRtpFile, kInputRefFile);
+  }
 }
 
 TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(TestNetworkStatistics)) {
@@ -523,14 +531,18 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(TestNetworkStatistics)) {
   // For Visual Studio 2012 and later, we will have to use the generic reference
   // file, rather than the windows-specific one.
   const std::string kNetworkStatRefFile = webrtc::test::ProjectRootPath() +
-      "resources/audio_coding/neteq_network_stats.dat";
+      "resources/audio_coding/neteq4_network_stats.dat";
 #else
   const std::string kNetworkStatRefFile =
-      webrtc::test::ResourcePath("audio_coding/neteq_network_stats", "dat");
+      webrtc::test::ResourcePath("audio_coding/neteq4_network_stats", "dat");
 #endif
   const std::string kRtcpStatRefFile =
-      webrtc::test::ResourcePath("audio_coding/neteq_rtcp_stats", "dat");
-  DecodeAndCheckStats(kInputRtpFile, kNetworkStatRefFile, kRtcpStatRefFile);
+      webrtc::test::ResourcePath("audio_coding/neteq4_rtcp_stats", "dat");
+  if (FLAGS_gen_ref) {
+    DecodeAndCheckStats(kInputRtpFile, "", "");
+  } else {
+    DecodeAndCheckStats(kInputRtpFile, kNetworkStatRefFile, kRtcpStatRefFile);
+  }
 }
 
 // TODO(hlundin): Re-enable test once the statistics interface is up and again.
diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp
index faa1ead7fb..38f9377044 100644
--- a/webrtc/modules/modules.gyp
+++ b/webrtc/modules/modules.gyp
@@ -90,6 +90,7 @@
             '<(rbe_components_path)/remote_bitrate_estimator_components.gyp:rbe_components',
             '<(DEPTH)/testing/gmock.gyp:gmock',
             '<(DEPTH)/testing/gtest.gyp:gtest',
+            '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
             '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
             '<(webrtc_root)/modules/video_coding/codecs/vp8/vp8.gyp:webrtc_vp8',
             '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
diff --git a/webrtc/tools/update_resources.py b/webrtc/tools/update_resources.py
index f68d7c6c47..d9ec564dbf 100755
--- a/webrtc/tools/update_resources.py
+++ b/webrtc/tools/update_resources.py
@@ -18,7 +18,7 @@ import tarfile
 import tempfile
 import urllib2
 
-DESIRED_VERSION = 16
+DESIRED_VERSION = 17
 REMOTE_URL_BASE = 'http://commondatastorage.googleapis.com/webrtc-resources'
 VERSION_FILENAME = 'webrtc-resources-version'
 FILENAME_PREFIX = 'webrtc-resources-'