diff --git a/webrtc/modules/audio_coding/neteq4/tools/rtp_generator.cc b/webrtc/modules/audio_coding/neteq4/tools/rtp_generator.cc
index 0ea28fb80..0b0e1ada4 100644
--- a/webrtc/modules/audio_coding/neteq4/tools/rtp_generator.cc
+++ b/webrtc/modules/audio_coding/neteq4/tools/rtp_generator.cc
@@ -33,9 +33,16 @@ uint32_t RtpGenerator::GetRtpHeader(uint8_t payload_type,
 
   uint32_t this_send_time = next_send_time_ms_;
   assert(samples_per_ms_ > 0);
-  next_send_time_ms_ += payload_length_samples / samples_per_ms_;
+  next_send_time_ms_ += ((1.0 + drift_factor_) * payload_length_samples) /
+      samples_per_ms_;
   return this_send_time;
 }
 
+void RtpGenerator::set_drift_factor(double factor) {
+  if (factor > -1.0) {
+    drift_factor_ = factor;
+  }
+}
+
 }  // namespace test
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/neteq4/tools/rtp_generator.h b/webrtc/modules/audio_coding/neteq4/tools/rtp_generator.h
index a2f885f31..ece7ef298 100644
--- a/webrtc/modules/audio_coding/neteq4/tools/rtp_generator.h
+++ b/webrtc/modules/audio_coding/neteq4/tools/rtp_generator.h
@@ -30,7 +30,8 @@ class RtpGenerator {
         timestamp_(start_timestamp),
         next_send_time_ms_(start_send_time_ms),
         ssrc_(ssrc),
-        samples_per_ms_(samples_per_ms) {
+        samples_per_ms_(samples_per_ms),
+        drift_factor_(0.0) {
   }
 
   // Writes the next RTP header to |rtp_header|, which will be of type
@@ -39,12 +40,15 @@ class RtpGenerator {
   uint32_t GetRtpHeader(uint8_t payload_type, size_t payload_length_samples,
                         WebRtcRTPHeader* rtp_header);
 
+  void set_drift_factor(double factor);
+
  private:
   uint16_t seq_number_;
   uint32_t timestamp_;
   uint32_t next_send_time_ms_;
   const uint32_t ssrc_;
   const int samples_per_ms_;
+  double drift_factor_;
   DISALLOW_COPY_AND_ASSIGN(RtpGenerator);
 };