diff --git a/webrtc/base/checks.h b/webrtc/base/checks.h
index 681361a3d2..e66c061186 100644
--- a/webrtc/base/checks.h
+++ b/webrtc/base/checks.h
@@ -220,7 +220,8 @@ class FatalMessage {
 // remainder is zero.
 template <typename T>
 inline T CheckedDivExact(T a, T b) {
-  RTC_CHECK_EQ(a % b, static_cast<T>(0));
+  RTC_CHECK_EQ(a % b, static_cast<T>(0)) << a << " is not evenly divisible by "
+                                         << b;
   return a / b;
 }
 
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
index 6170d187ba..b394c59156 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
@@ -472,6 +472,11 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
   if (!HaveValidEncoder("Process"))
     return -1;
 
+  if(!first_frame_) {
+    RTC_DCHECK_GT(input_data.input_timestamp, last_timestamp_)
+        << "Time should not move backwards";
+  }
+
   // Scale the timestamp to the codec's RTP timestamp rate.
   uint32_t rtp_timestamp =
       first_frame_ ? input_data.input_timestamp
@@ -752,7 +757,8 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
     expected_codec_ts_ = in_frame.timestamp_;
     first_10ms_data_ = true;
   } else if (in_frame.timestamp_ != expected_in_ts_) {
-    // TODO(turajs): Do we need a warning here.
+    LOG(LS_WARNING) << "Unexpected input timestamp: " << in_frame.timestamp_
+                    << ", expected: " << expected_in_ts_;
     expected_codec_ts_ +=
         (in_frame.timestamp_ - expected_in_ts_) *
         static_cast<uint32_t>(
@@ -764,9 +770,19 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
 
   if (!down_mix && !resample) {
     // No pre-processing is required.
+    if (expected_in_ts_ == expected_codec_ts_) {
+      // If we've never resampled, we can use the input frame as-is
+      *ptr_out = &in_frame;
+    } else {
+      // Otherwise we'll need to alter the timestamp. Since in_frame is const,
+      // we'll have to make a copy of it.
+      preprocess_frame_.CopyFrom(in_frame);
+      preprocess_frame_.timestamp_ = expected_codec_ts_;
+      *ptr_out = &preprocess_frame_;
+    }
+
     expected_in_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
     expected_codec_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
-    *ptr_out = &in_frame;
     return 0;
   }
 
diff --git a/webrtc/modules/audio_coding/test/TestVADDTX.cc b/webrtc/modules/audio_coding/test/TestVADDTX.cc
index 4f53e47cac..541dfc3c45 100644
--- a/webrtc/modules/audio_coding/test/TestVADDTX.cc
+++ b/webrtc/modules/audio_coding/test/TestVADDTX.cc
@@ -101,12 +101,11 @@ void TestVadDtx::Run(std::string in_filename, int frequency, int channels,
   }
 
   uint16_t frame_size_samples = in_file.PayloadLength10Ms();
-  uint32_t time_stamp = 0x12345678;
   AudioFrame audio_frame;
   while (!in_file.EndOfFile()) {
     in_file.Read10MsData(audio_frame);
-    audio_frame.timestamp_ = time_stamp;
-    time_stamp += frame_size_samples;
+    audio_frame.timestamp_ = time_stamp_;
+    time_stamp_ += frame_size_samples;
     EXPECT_GE(acm_send_->Add10MsData(audio_frame), 0);
     bool muted;
     acm_receive_->PlayoutData10Ms(kOutputFreqHz, &audio_frame, &muted);
diff --git a/webrtc/modules/audio_coding/test/TestVADDTX.h b/webrtc/modules/audio_coding/test/TestVADDTX.h
index 893babc4e0..b7e987137c 100644
--- a/webrtc/modules/audio_coding/test/TestVADDTX.h
+++ b/webrtc/modules/audio_coding/test/TestVADDTX.h
@@ -72,6 +72,7 @@ class TestVadDtx : public ACMTest {
   std::unique_ptr<AudioCodingModule> acm_receive_;
   std::unique_ptr<Channel> channel_;
   std::unique_ptr<ActivityMonitor> monitor_;
+  uint32_t time_stamp_ = 0x12345678;
 };
 
 // TestWebRtcVadDtx is to verify that the WebRTC VAD/DTX perform as they should.