diff --git a/experiments/field_trials.py b/experiments/field_trials.py
index 5a95331ef8..be0f62db22 100755
--- a/experiments/field_trials.py
+++ b/experiments/field_trials.py
@@ -475,9 +475,6 @@ POLICY_EXEMPT_FIELD_TRIALS: FrozenSet[FieldTrial] = frozenset([
     FieldTrial('WebRTC-Audio-OpusBitrateMultipliers',
                42221139,
                date(2024, 4, 1)),
-    FieldTrial('WebRTC-Audio-OpusPlcUsePrevDecodedSamples',
-               143582588,
-               date(2024, 4, 1)),
     FieldTrial('WebRTC-Audio-Red-For-Opus',
                42221750,
                date(2024, 4, 1)),
@@ -878,7 +875,7 @@ POLICY_EXEMPT_FIELD_TRIALS: FrozenSet[FieldTrial] = frozenset([
 ])  # yapf: disable
 
 POLICY_EXEMPT_FIELD_TRIALS_DIGEST: str = \
-    '95050aa4a628b16d1220b3674b543c035c24cb6c'
+    'e750429cb042c02921a227fa18291f2cea23ea3b'
 
 REGISTERED_FIELD_TRIALS: FrozenSet[FieldTrial] = ACTIVE_FIELD_TRIALS.union(
     POLICY_EXEMPT_FIELD_TRIALS)
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 78c0417da9..a93d434c8e 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -545,7 +545,6 @@ rtc_library("webrtc_opus_wrapper") {
     "../../api:array_view",
     "../../rtc_base:checks",
     "../../rtc_base:ignore_wundef",
-    "../../system_wrappers:field_trial",
   ]
 
   if (rtc_build_opus) {
diff --git a/modules/audio_coding/codecs/opus/opus_inst.h b/modules/audio_coding/codecs/opus/opus_inst.h
index 34bb7710e5..ddf4396256 100644
--- a/modules/audio_coding/codecs/opus/opus_inst.h
+++ b/modules/audio_coding/codecs/opus/opus_inst.h
@@ -31,8 +31,6 @@ struct WebRtcOpusEncInst {
 struct WebRtcOpusDecInst {
   OpusDecoder* decoder;
   OpusMSDecoder* multistream_decoder;
-  int prev_decoded_samples;
-  bool plc_use_prev_decoded_samples;
   size_t channels;
   int in_dtx_mode;
   int sample_rate_hz;
diff --git a/modules/audio_coding/codecs/opus/opus_interface.cc b/modules/audio_coding/codecs/opus/opus_interface.cc
index e5ca678120..1257f25c7c 100644
--- a/modules/audio_coding/codecs/opus/opus_interface.cc
+++ b/modules/audio_coding/codecs/opus/opus_interface.cc
@@ -15,7 +15,6 @@
 
 #include "api/array_view.h"
 #include "rtc_base/checks.h"
-#include "system_wrappers/include/field_trial.h"
 
 enum {
 #if WEBRTC_OPUS_SUPPORT_120MS_PTIME
@@ -35,9 +34,6 @@ enum {
   kWebRtcOpusPlcFrameSizeMs = 10,
 };
 
-constexpr char kPlcUsePrevDecodedSamplesFieldTrial[] =
-    "WebRTC-Audio-OpusPlcUsePrevDecodedSamples";
-
 static int FrameSizePerChannel(int frame_size_ms, int sample_rate_hz) {
   RTC_DCHECK_GT(frame_size_ms, 0);
   RTC_DCHECK_EQ(frame_size_ms % 10, 0);
@@ -51,11 +47,6 @@ static int MaxFrameSizePerChannel(int sample_rate_hz) {
   return FrameSizePerChannel(kWebRtcOpusMaxDecodeFrameSizeMs, sample_rate_hz);
 }
 
-// Default sample count per channel.
-static int DefaultFrameSizePerChannel(int sample_rate_hz) {
-  return FrameSizePerChannel(20, sample_rate_hz);
-}
-
 int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst,
                                  size_t channels,
                                  int32_t application,
@@ -394,12 +385,6 @@ int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst,
       // Creation of memory all ok.
       state->channels = channels;
       state->sample_rate_hz = sample_rate_hz;
-      state->plc_use_prev_decoded_samples =
-          webrtc::field_trial::IsEnabled(kPlcUsePrevDecodedSamplesFieldTrial);
-      if (state->plc_use_prev_decoded_samples) {
-        state->prev_decoded_samples =
-            DefaultFrameSizePerChannel(state->sample_rate_hz);
-      }
       state->in_dtx_mode = 0;
       *inst = state;
       return 0;
@@ -438,12 +423,6 @@ int16_t WebRtcOpus_MultistreamDecoderCreate(
       // Creation of memory all ok.
       state->channels = channels;
       state->sample_rate_hz = 48000;
-      state->plc_use_prev_decoded_samples =
-          webrtc::field_trial::IsEnabled(kPlcUsePrevDecodedSamplesFieldTrial);
-      if (state->plc_use_prev_decoded_samples) {
-        state->prev_decoded_samples =
-            DefaultFrameSizePerChannel(state->sample_rate_hz);
-      }
       state->in_dtx_mode = 0;
       *inst = state;
       return 0;
@@ -542,17 +521,6 @@ static int DecodePlc(OpusDecInst* inst, int16_t* decoded) {
   int plc_samples =
       FrameSizePerChannel(kWebRtcOpusPlcFrameSizeMs, inst->sample_rate_hz);
 
-  if (inst->plc_use_prev_decoded_samples) {
-    /* The number of samples we ask for is `number_of_lost_frames` times
-     * `prev_decoded_samples_`. Limit the number of samples to maximum
-     * `MaxFrameSizePerChannel()`. */
-    plc_samples = inst->prev_decoded_samples;
-    const int max_samples_per_channel =
-        MaxFrameSizePerChannel(inst->sample_rate_hz);
-    plc_samples = plc_samples <= max_samples_per_channel
-                      ? plc_samples
-                      : max_samples_per_channel;
-  }
   decoded_samples =
       DecodeNative(inst, NULL, 0, plc_samples, decoded, &audio_type, 0);
   if (decoded_samples < 0) {
@@ -581,11 +549,6 @@ int WebRtcOpus_Decode(OpusDecInst* inst,
     return -1;
   }
 
-  if (inst->plc_use_prev_decoded_samples) {
-    /* Update decoded sample memory, to be used by the PLC in case of losses. */
-    inst->prev_decoded_samples = decoded_samples;
-  }
-
   return decoded_samples;
 }
 
@@ -639,16 +602,6 @@ int WebRtcOpus_DurationEst(OpusDecInst* inst,
 }
 
 int WebRtcOpus_PlcDuration(OpusDecInst* inst) {
-  if (inst->plc_use_prev_decoded_samples) {
-    /* The number of samples we ask for is `number_of_lost_frames` times
-     * `prev_decoded_samples_`. Limit the number of samples to maximum
-     * `MaxFrameSizePerChannel()`. */
-    const int plc_samples = inst->prev_decoded_samples;
-    const int max_samples_per_channel =
-        MaxFrameSizePerChannel(inst->sample_rate_hz);
-    return plc_samples <= max_samples_per_channel ? plc_samples
-                                                  : max_samples_per_channel;
-  }
   return FrameSizePerChannel(kWebRtcOpusPlcFrameSizeMs, inst->sample_rate_hz);
 }