diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 6836c7e59a..85e3690c9d 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -103,9 +103,7 @@ NetEqImpl::NetEqImpl(const NetEq::Config& config,
       playout_mode_(config.playout_mode),
       enable_fast_accelerate_(config.enable_fast_accelerate),
       nack_enabled_(false),
-      enable_muted_state_(config.enable_muted_state),
-      use_dtx_delay_fix_(
-          field_trial::IsEnabled("WebRTC-NetEqOpusDtxDelayFix")) {
+      enable_muted_state_(config.enable_muted_state) {
   RTC_LOG(LS_INFO) << "NetEq config: " << config.ToString();
   int fs = config.sample_rate_hz;
   if (fs != 8000 && fs != 16000 && fs != 32000 && fs != 48000) {
@@ -877,9 +875,8 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame, bool* muted) {
   // This is the criterion that we did decode some data through the speech
   // decoder, and the operation resulted in comfort noise.
   const bool codec_internal_sid_frame =
-      use_dtx_delay_fix_ ? (speech_type == AudioDecoder::kComfortNoise &&
-                            start_num_packets > packet_list.size())
-                         : (speech_type == AudioDecoder::kComfortNoise);
+      (speech_type == AudioDecoder::kComfortNoise &&
+       start_num_packets > packet_list.size());
 
   if (sid_frame_available || codec_internal_sid_frame) {
     // Start a new stopwatch since we are decoding a new CNG packet.
diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h
index a922bcdcd6..bdeb0206d6 100644
--- a/modules/audio_coding/neteq/neteq_impl.h
+++ b/modules/audio_coding/neteq/neteq_impl.h
@@ -440,7 +440,6 @@ class NetEqImpl : public webrtc::NetEq {
   std::unique_ptr<TickTimer::Stopwatch> generated_noise_stopwatch_
       RTC_GUARDED_BY(crit_sect_);
   std::vector<uint32_t> last_decoded_timestamps_ RTC_GUARDED_BY(crit_sect_);
-  const bool use_dtx_delay_fix_ RTC_GUARDED_BY(crit_sect_);
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index ef31f4b56e..ca93cf5533 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -529,20 +529,6 @@ TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) {
                    FLAG_gen_ref);
 }
 
-// This test fixture is identical to NetEqDecodingTest, except that it enables
-// the WebRTC-NetEqOpusDtxDelayFix field trial.
-// TODO(bugs.webrtc.org/8488): When the field trial is over and the feature is
-// default enabled, remove this fixture class and let the
-// TestOpusDtxBitExactness test build directly on NetEqDecodingTest.
-class NetEqDecodingTestWithOpusDtxFieldTrial : public NetEqDecodingTest {
- public:
-  NetEqDecodingTestWithOpusDtxFieldTrial()
-      : override_field_trials_("WebRTC-NetEqOpusDtxDelayFix/Enabled/") {}
-
- private:
-  test::ScopedFieldTrials override_field_trials_;
-};
-
 #if !defined(WEBRTC_IOS) &&                                         \
     defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) &&                      \
     defined(WEBRTC_CODEC_OPUS)
@@ -550,7 +536,7 @@ class NetEqDecodingTestWithOpusDtxFieldTrial : public NetEqDecodingTest {
 #else
 #define MAYBE_TestOpusDtxBitExactness DISABLED_TestOpusDtxBitExactness
 #endif
-TEST_F(NetEqDecodingTestWithOpusDtxFieldTrial, MAYBE_TestOpusDtxBitExactness) {
+TEST_F(NetEqDecodingTest, MAYBE_TestOpusDtxBitExactness) {
   const std::string input_rtp_file =
       webrtc::test::ResourcePath("audio_coding/neteq_opus_dtx", "rtp");