diff --git a/modules/audio_coding/codecs/opus/audio_decoder_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_decoder_opus_unittest.cc
index 4610a784ce..71f0d38b25 100644
--- a/modules/audio_coding/codecs/opus/audio_decoder_opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/audio_decoder_opus_unittest.cc
@@ -27,12 +27,15 @@
 #include "rtc_base/buffer.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/random.h"
+#include "test/explicit_key_value_config.h"
 #include "test/gtest.h"
 #include "test/testsupport/file_utils.h"
 
 namespace webrtc {
 namespace {
 
+using test::ExplicitKeyValueConfig;
+
 using DecodeResult = ::webrtc::AudioDecoder::EncodedAudioFrame::DecodeResult;
 using ParseResult = ::webrtc::AudioDecoder::ParseResult;
 
@@ -225,7 +228,8 @@ TEST(AudioDecoderOpusTest, MonoEncoderStereoDecoderOutputsTrivialStereo) {
   }
 }
 
-TEST(AudioDecoderOpusTest, MonoEncoderStereoDecoderOutputsNonTrivialStereoDtx) {
+TEST(AudioDecoderOpusTest,
+     MonoEncoderStereoDecoderOutputsTrivialStereoComfortNoise) {
   const Environment env = EnvironmentFactory().Create();
   // Create a mono encoder.
   const AudioEncoderOpusConfig encoder_config =
@@ -259,12 +263,45 @@ TEST(AudioDecoderOpusTest, MonoEncoderStereoDecoderOutputsNonTrivialStereoDtx) {
   // Make sure that comfort noise is not a muted frame.
   ASSERT_FALSE(IsZeroedFrame(decoded_view));
 
-  // TODO: https://issues.webrtc.org/376493209 - When fixed, expect true below.
-  EXPECT_FALSE(IsTrivialStereo(decoded_view));
+  EXPECT_TRUE(IsTrivialStereo(decoded_view));
+}
+
+TEST(AudioDecoderOpusTest, MonoEncoderStereoDecoderOutputsTrivialStereoPlc) {
+  const ExplicitKeyValueConfig trials("WebRTC-Audio-OpusGeneratePlc/Enabled/");
+  EnvironmentFactory env_factory;
+  env_factory.Set(&trials);
+  const Environment env = env_factory.Create();
+  // Create a mono encoder.
+  const AudioEncoderOpusConfig encoder_config =
+      GetEncoderConfig(/*num_channels=*/1, /*dtx_enabled=*/false);
+  AudioEncoderOpusImpl encoder(env, encoder_config, kPayloadType);
+  // Create a stereo decoder.
+  constexpr size_t kDecoderNumChannels = 2;
+  AudioDecoderOpusImpl decoder(env.field_trials(), kDecoderNumChannels,
+                               kSampleRateHz);
+
+  uint32_t rtp_timestamp = 0xFFFu;
+  uint32_t timestamp = 0;
+  // Feed the encoder with speech.
+  EncodeDecodeSpeech(encoder, decoder, rtp_timestamp, timestamp,
+                     /*max_frames=*/100);
+
+  // Generate packet loss concealment.
+  rtc::BufferT<int16_t> concealment_audio;
+  constexpr int kIgnored = 123;
+  decoder.GeneratePlc(/*requested_samples_per_channel=*/kIgnored,
+                      &concealment_audio);
+  RTC_CHECK_GT(concealment_audio.size(), 0);
+  rtc::ArrayView<const int16_t> decoded_view(concealment_audio.data(),
+                                             concealment_audio.size());
+  // Make sure that packet loss concealment is not a muted frame.
+  ASSERT_FALSE(IsZeroedFrame(decoded_view));
+
+  EXPECT_TRUE(IsTrivialStereo(decoded_view));
 }
 
 TEST(AudioDecoderOpusTest,
-     StereoEncoderStereoDecoderOutputsNonTrivialStereoDtx) {
+     StereoEncoderStereoDecoderOutputsNonTrivialStereoComfortNoise) {
   const Environment env = EnvironmentFactory().Create();
   // Create a stereo encoder.
   const AudioEncoderOpusConfig encoder_config =
@@ -301,4 +338,39 @@ TEST(AudioDecoderOpusTest,
   EXPECT_FALSE(IsTrivialStereo(decoded_view));
 }
 
+TEST(AudioDecoderOpusTest,
+     StereoEncoderStereoDecoderOutputsNonTrivialStereoPlc) {
+  const ExplicitKeyValueConfig trials("WebRTC-Audio-OpusGeneratePlc/Enabled/");
+  EnvironmentFactory env_factory;
+  env_factory.Set(&trials);
+  const Environment env = env_factory.Create();
+  // Create a stereo encoder.
+  const AudioEncoderOpusConfig encoder_config =
+      GetEncoderConfig(/*num_channels=*/2, /*dtx_enabled=*/false);
+  AudioEncoderOpusImpl encoder(env, encoder_config, kPayloadType);
+  // Create a stereo decoder.
+  constexpr size_t kDecoderNumChannels = 2;
+  AudioDecoderOpusImpl decoder(env.field_trials(), kDecoderNumChannels,
+                               kSampleRateHz);
+
+  uint32_t rtp_timestamp = 0xFFFu;
+  uint32_t timestamp = 0;
+  // Feed the encoder with speech.
+  EncodeDecodeSpeech(encoder, decoder, rtp_timestamp, timestamp,
+                     /*max_frames=*/100);
+
+  // Generate packet loss concealment.
+  rtc::BufferT<int16_t> concealment_audio;
+  constexpr int kIgnored = 123;
+  decoder.GeneratePlc(/*requested_samples_per_channel=*/kIgnored,
+                      &concealment_audio);
+  RTC_CHECK_GT(concealment_audio.size(), 0);
+  rtc::ArrayView<const int16_t> decoded_view(concealment_audio.data(),
+                                             concealment_audio.size());
+  // Make sure that packet loss concealment is not a muted frame.
+  ASSERT_FALSE(IsZeroedFrame(decoded_view));
+
+  EXPECT_FALSE(IsTrivialStereo(decoded_view));
+}
+
 }  // namespace webrtc
diff --git a/modules/audio_coding/codecs/opus/opus_inst.h b/modules/audio_coding/codecs/opus/opus_inst.h
index ddf4396256..f7b45ea69b 100644
--- a/modules/audio_coding/codecs/opus/opus_inst.h
+++ b/modules/audio_coding/codecs/opus/opus_inst.h
@@ -34,6 +34,8 @@ struct WebRtcOpusDecInst {
   size_t channels;
   int in_dtx_mode;
   int sample_rate_hz;
+  // TODO: https://issues.webrtc.org/376493209 - Remove when libopus gets fixed.
+  int last_packet_num_channels;
 };
 
 #endif  // MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
diff --git a/modules/audio_coding/codecs/opus/opus_interface.cc b/modules/audio_coding/codecs/opus/opus_interface.cc
index 4b661c8d0e..565f803741 100644
--- a/modules/audio_coding/codecs/opus/opus_interface.cc
+++ b/modules/audio_coding/codecs/opus/opus_interface.cc
@@ -396,6 +396,7 @@ int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst,
       state->channels = channels;
       state->sample_rate_hz = sample_rate_hz;
       state->in_dtx_mode = 0;
+      state->last_packet_num_channels = channels;
       *inst = state;
       return 0;
     }
@@ -545,21 +546,39 @@ int WebRtcOpus_Decode(OpusDecInst* inst,
                       size_t encoded_bytes,
                       int16_t* decoded,
                       int16_t* audio_type) {
-  int decoded_samples;
+  int decoded_samples_per_channel;
 
   if (encoded_bytes == 0) {
     *audio_type = DetermineAudioType(inst, encoded_bytes);
-    decoded_samples = DecodePlc(inst, decoded);
+    decoded_samples_per_channel = DecodePlc(inst, decoded);
+
+    // TODO: https://issues.webrtc.org/376493209 - When fixed, remove block
+    // below.
+    if (inst->channels == 2 && inst->last_packet_num_channels == 1) {
+      // Stereo decoding is enabled and the last observed packet to decode
+      // encoded mono audio. In this case, Opus generates non-trivial stereo
+      // audio. Since this is unwanted, copy the left channel into the right
+      // one.
+      for (int i = 0; i < decoded_samples_per_channel << 1; i += 2) {
+        decoded[i + 1] = decoded[i];
+      }
+    }
   } else {
-    decoded_samples = DecodeNative(inst, encoded, encoded_bytes,
-                                   MaxFrameSizePerChannel(inst->sample_rate_hz),
-                                   decoded, audio_type, 0);
+    decoded_samples_per_channel = DecodeNative(
+        inst, encoded, encoded_bytes,
+        MaxFrameSizePerChannel(inst->sample_rate_hz), decoded, audio_type, 0);
+
+    // TODO: https://issues.webrtc.org/376493209 - When fixed, remove block
+    // below.
+    const int num_channels = opus_packet_get_nb_channels(encoded);
+    RTC_DCHECK(num_channels == 1 || num_channels == 2);
+    inst->last_packet_num_channels = num_channels;
   }
-  if (decoded_samples < 0) {
+  if (decoded_samples_per_channel < 0) {
     return -1;
   }
 
-  return decoded_samples;
+  return decoded_samples_per_channel;
 }
 
 int WebRtcOpus_DecodeFec(OpusDecInst* inst,