diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index ff5d71b605..220e96f1b7 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -357,7 +357,7 @@ AudioEncoderOpusImpl::AudioEncoderOpusImpl(
     : payload_type_(payload_type),
       send_side_bwe_with_overhead_(
           webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
-      use_stable_target_for_adaptation_(webrtc::field_trial::IsEnabled(
+      use_stable_target_for_adaptation_(!webrtc::field_trial::IsDisabled(
           "WebRTC-Audio-StableTargetAdaptation")),
       adjust_bandwidth_(
           webrtc::field_trial::IsEnabled("WebRTC-AdjustOpusBandwidth")),
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index 1c73eaf717..1cbc4a3ff7 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -254,6 +254,8 @@ TEST_P(AudioEncoderOpusTest,
 
 TEST_P(AudioEncoderOpusTest,
        InvokeAudioNetworkAdaptorOnReceivedUplinkBandwidth) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Audio-StableTargetAdaptation/Disabled/");
   auto states = CreateCodec(sample_rate_hz_, 2);
   states->encoder->EnableAudioNetworkAdaptor("", nullptr);
 
@@ -275,6 +277,28 @@ TEST_P(AudioEncoderOpusTest,
   CheckEncoderRuntimeConfig(states->encoder.get(), config);
 }
 
+TEST_P(AudioEncoderOpusTest,
+       InvokeAudioNetworkAdaptorOnReceivedUplinkAllocation) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
+
+  auto config = CreateEncoderRuntimeConfig();
+  EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
+      .WillOnce(Return(config));
+
+  BitrateAllocationUpdate update;
+  update.target_bitrate = DataRate::BitsPerSec(30000);
+  update.stable_target_bitrate = DataRate::BitsPerSec(20000);
+  update.bwe_period = TimeDelta::Millis(200);
+  EXPECT_CALL(*states->mock_audio_network_adaptor,
+              SetTargetAudioBitrate(update.target_bitrate.bps()));
+  EXPECT_CALL(*states->mock_audio_network_adaptor,
+              SetUplinkBandwidth(update.stable_target_bitrate.bps()));
+  states->encoder->OnReceivedUplinkAllocation(update);
+
+  CheckEncoderRuntimeConfig(states->encoder.get(), config);
+}
+
 TEST_P(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnReceivedRtt) {
   auto states = CreateCodec(sample_rate_hz_, 2);
   states->encoder->EnableAudioNetworkAdaptor("", nullptr);
@@ -501,6 +525,8 @@ TEST_P(AudioEncoderOpusTest, EmptyConfigDoesNotAffectEncoderSettings) {
 }
 
 TEST_P(AudioEncoderOpusTest, UpdateUplinkBandwidthInAudioNetworkAdaptor) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Audio-StableTargetAdaptation/Disabled/");
   auto states = CreateCodec(sample_rate_hz_, 2);
   states->encoder->EnableAudioNetworkAdaptor("", nullptr);
   const size_t opus_rate_khz = rtc::CheckedDivExact(sample_rate_hz_, 1000);