diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
index 925e99c5d6..f8bacf8dc7 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
@@ -191,21 +191,6 @@ int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
              audio_frame->num_channels_);
 
   call_stats_.DecodedByNetEq(audio_frame->speech_type_);
-
-  // Computes the RTP timestamp of the first sample in |audio_frame| from
-  // |GetPlayoutTimestamp|, which is the timestamp of the last sample of
-  // |audio_frame|.
-  // TODO(henrik.lundin) Move setting of audio_frame->timestamp_ inside NetEq.
-  rtc::Optional<uint32_t> playout_timestamp = GetPlayoutTimestamp();
-  if (playout_timestamp) {
-    audio_frame->timestamp_ =
-        *playout_timestamp -
-        static_cast<uint32_t>(audio_frame->samples_per_channel_);
-  } else {
-    // Remain 0 until we have a valid |playout_timestamp|.
-    audio_frame->timestamp_ = 0;
-  }
-
   return 0;
 }
 
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc b/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
index 969ff40bc2..c39a7cc797 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
@@ -326,6 +326,9 @@ class AcmReceiverTestFaxModeOldApi : public AcmReceiverTestOldApi {
         rtc::CheckedDivExact(5 * output_sample_rate_hz, 8000);
 
     AudioFrame frame;
+    EXPECT_EQ(0, receiver_->GetAudio(output_sample_rate_hz, &frame));
+    // Expect timestamp = 0 before first packet is inserted.
+    EXPECT_EQ(0u, frame.timestamp_);
     for (int i = 0; i < 5; ++i) {
       InsertOnePacketOfSilence(codec.id);
       for (int k = 0; k < num_10ms_frames; ++k) {
diff --git a/webrtc/modules/audio_coding/neteq/include/neteq.h b/webrtc/modules/audio_coding/neteq/include/neteq.h
index 5b52424bee..89b0c54324 100644
--- a/webrtc/modules/audio_coding/neteq/include/neteq.h
+++ b/webrtc/modules/audio_coding/neteq/include/neteq.h
@@ -246,7 +246,7 @@ class NetEq {
 
   // Returns the RTP timestamp for the last sample delivered by GetAudio().
   // The return value will be empty if no valid timestamp is available.
-  virtual rtc::Optional<uint32_t> GetPlayoutTimestamp() = 0;
+  virtual rtc::Optional<uint32_t> GetPlayoutTimestamp() const = 0;
 
   // Returns the sample rate in Hz of the audio produced in the last GetAudio
   // call. If GetAudio has not been called yet, the configured sample rate
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index b4dfc9986d..40a031421e 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -401,7 +401,7 @@ void NetEqImpl::DisableVad() {
   vad_->Disable();
 }
 
-rtc::Optional<uint32_t> NetEqImpl::GetPlayoutTimestamp() {
+rtc::Optional<uint32_t> NetEqImpl::GetPlayoutTimestamp() const {
   rtc::CritScope lock(&crit_sect_);
   if (first_packet_) {
     // We don't have a valid RTP timestamp until we have decoded our first
@@ -966,6 +966,15 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame) {
     // Use dead reckoning to estimate the |playout_timestamp_|.
     playout_timestamp_ += static_cast<uint32_t>(output_size_samples_);
   }
+  // Set the timestamp in the audio frame to zero before the first packet has
+  // been inserted. Otherwise, subtract the frame size in samples to get the
+  // timestamp of the first sample in the frame (playout_timestamp_ is the
+  // last + 1).
+  audio_frame->timestamp_ =
+      first_packet_
+          ? 0
+          : timestamp_scaler_->ToExternal(playout_timestamp_) -
+                static_cast<uint32_t>(audio_frame->samples_per_channel_);
 
   if (decode_return_value) return decode_return_value;
   return return_value;
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.h b/webrtc/modules/audio_coding/neteq/neteq_impl.h
index 737efd46da..75055a7b47 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.h
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.h
@@ -160,7 +160,7 @@ class NetEqImpl : public webrtc::NetEq {
   // Disables post-decode VAD.
   void DisableVad() override;
 
-  rtc::Optional<uint32_t> GetPlayoutTimestamp() override;
+  rtc::Optional<uint32_t> GetPlayoutTimestamp() const override;
 
   int last_output_sample_rate_hz() const override;