diff --git a/api/audio_codecs/audio_decoder_factory_template.h b/api/audio_codecs/audio_decoder_factory_template.h
index 47f47b5b38..6490f4c2a9 100644
--- a/api/audio_codecs/audio_decoder_factory_template.h
+++ b/api/audio_codecs/audio_decoder_factory_template.h
@@ -58,11 +58,16 @@ struct Helper<T, Ts...> {
                   "absl::optional<T::Config>");
     return opt_config ? true : Helper<Ts...>::IsSupportedDecoder(format);
   }
+  // TODO: bugs.webrtc.org/356878416 - Migrate to `Create`
   static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
       const SdpAudioFormat& format,
       absl::optional<AudioCodecPairId> codec_pair_id) {
     auto opt_config = T::SdpToConfig(format);
-    return opt_config ? T::MakeAudioDecoder(*opt_config, codec_pair_id)
+    return opt_config ?
+#pragma clang diagnostic push
+#pragma clang diagnostic ignored "-Wdeprecated-declarations"
+                      T::MakeAudioDecoder(*opt_config, codec_pair_id)
+#pragma clang diagnostic pop
                       : Helper<Ts...>::MakeAudioDecoder(format, codec_pair_id);
   }
 };
diff --git a/api/audio_codecs/opus/BUILD.gn b/api/audio_codecs/opus/BUILD.gn
index 2c0dbefc7f..8c673116de 100644
--- a/api/audio_codecs/opus/BUILD.gn
+++ b/api/audio_codecs/opus/BUILD.gn
@@ -67,6 +67,8 @@ rtc_library("audio_decoder_opus") {
     "../../../modules/audio_coding:webrtc_opus",
     "../../../rtc_base:checks",
     "../../../rtc_base/system:rtc_export",
+    "../../environment",
+    "../../transport:field_trial_based_config",
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
diff --git a/api/audio_codecs/opus/audio_decoder_opus.cc b/api/audio_codecs/opus/audio_decoder_opus.cc
index 3c8635d6c8..e1ba301cbd 100644
--- a/api/audio_codecs/opus/audio_decoder_opus.cc
+++ b/api/audio_codecs/opus/audio_decoder_opus.cc
@@ -22,6 +22,7 @@
 #include "api/audio_codecs/audio_decoder.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/field_trials_view.h"
+#include "api/transport/field_trial_based_config.h"
 #include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
 #include "rtc_base/checks.h"
 
@@ -79,6 +80,17 @@ void AudioDecoderOpus::AppendSupportedDecoders(
   specs->push_back({std::move(opus_format), opus_info});
 }
 
+std::unique_ptr<AudioDecoder> AudioDecoderOpus::MakeAudioDecoder(
+    const Environment& env,
+    Config config) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  return std::make_unique<AudioDecoderOpusImpl>(
+      env.field_trials(), config.num_channels, config.sample_rate_hz);
+}
+
 std::unique_ptr<AudioDecoder> AudioDecoderOpus::MakeAudioDecoder(
     Config config,
     absl::optional<AudioCodecPairId> /*codec_pair_id*/,
@@ -87,8 +99,13 @@ std::unique_ptr<AudioDecoder> AudioDecoderOpus::MakeAudioDecoder(
     RTC_DCHECK_NOTREACHED();
     return nullptr;
   }
-  return std::make_unique<AudioDecoderOpusImpl>(config.num_channels,
-                                                config.sample_rate_hz);
+  if (field_trials == nullptr) {
+    return std::make_unique<AudioDecoderOpusImpl>(
+        FieldTrialBasedConfig(), config.num_channels, config.sample_rate_hz);
+  } else {
+    return std::make_unique<AudioDecoderOpusImpl>(
+        *field_trials, config.num_channels, config.sample_rate_hz);
+  }
 }
 
 }  // namespace webrtc
diff --git a/api/audio_codecs/opus/audio_decoder_opus.h b/api/audio_codecs/opus/audio_decoder_opus.h
index 138c0377df..b4d84cb2d5 100644
--- a/api/audio_codecs/opus/audio_decoder_opus.h
+++ b/api/audio_codecs/opus/audio_decoder_opus.h
@@ -18,6 +18,7 @@
 #include "api/audio_codecs/audio_codec_pair_id.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "api/audio_codecs/audio_format.h"
+#include "api/environment/environment.h"
 #include "api/field_trials_view.h"
 #include "rtc_base/system/rtc_export.h"
 
@@ -33,10 +34,16 @@ struct RTC_EXPORT AudioDecoderOpus {
   };
   static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
   static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
-  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+
+  [[deprecated("bugs.webrtc.org/356878416")]]  //
+  static std::unique_ptr<AudioDecoder>
+  MakeAudioDecoder(
       Config config,
       absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
       const FieldTrialsView* field_trials = nullptr);
+
+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(const Environment& env,
+                                                        Config config);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index a93d434c8e..cb98266510 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -485,7 +485,6 @@ rtc_library("webrtc_opus") {
     "../../rtc_base:safe_minmax",
     "../../rtc_base:stringutils",
     "../../rtc_base:timeutils",
-    "../../system_wrappers:field_trial",
     "//third_party/abseil-cpp/absl/memory",
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/strings:string_view",
@@ -1207,6 +1206,7 @@ if (rtc_include_tests) {
                "../../api/environment:environment_factory",
                "../../common_audio",
                "../../rtc_base/system:arch",
+               "../../test:explicit_key_value_config",
                "../../test:fileutils",
                "../../test:test_main",
                "../../test:test_support",
diff --git a/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
index 0f53409f48..7d5e3cfd60 100644
--- a/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
@@ -15,17 +15,18 @@
 
 #include "absl/types/optional.h"
 #include "api/array_view.h"
+#include "api/field_trials_view.h"
 #include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
 #include "rtc_base/checks.h"
-#include "system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 
-AudioDecoderOpusImpl::AudioDecoderOpusImpl(size_t num_channels,
+AudioDecoderOpusImpl::AudioDecoderOpusImpl(const FieldTrialsView& field_trials,
+                                           size_t num_channels,
                                            int sample_rate_hz)
     : channels_(num_channels),
       sample_rate_hz_(sample_rate_hz),
-      generate_plc_(field_trial::IsEnabled("WebRTC-Audio-OpusGeneratePlc")) {
+      generate_plc_(field_trials.IsEnabled("WebRTC-Audio-OpusGeneratePlc")) {
   RTC_DCHECK(num_channels == 1 || num_channels == 2);
   RTC_DCHECK(sample_rate_hz == 16000 || sample_rate_hz == 48000);
   const int error =
diff --git a/modules/audio_coding/codecs/opus/audio_decoder_opus.h b/modules/audio_coding/codecs/opus/audio_decoder_opus.h
index 2dd62fd4ee..55dbf514f5 100644
--- a/modules/audio_coding/codecs/opus/audio_decoder_opus.h
+++ b/modules/audio_coding/codecs/opus/audio_decoder_opus.h
@@ -17,6 +17,7 @@
 #include <vector>
 
 #include "api/audio_codecs/audio_decoder.h"
+#include "api/field_trials_view.h"
 #include "modules/audio_coding/codecs/opus/opus_interface.h"
 #include "rtc_base/buffer.h"
 
@@ -24,8 +25,10 @@ namespace webrtc {
 
 class AudioDecoderOpusImpl final : public AudioDecoder {
  public:
-  explicit AudioDecoderOpusImpl(size_t num_channels,
-                                int sample_rate_hz = 48000);
+  explicit AudioDecoderOpusImpl(const FieldTrialsView& field_trails,
+                                size_t num_channels,
+                                int sample_rate_hz);
+
   ~AudioDecoderOpusImpl() override;
 
   AudioDecoderOpusImpl(const AudioDecoderOpusImpl&) = delete;
diff --git a/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc b/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
index 423d389db2..85b9293029 100644
--- a/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
+++ b/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
@@ -122,7 +122,7 @@ TEST(BandwidthAdaptationTest, BandwidthAdaptationTest) {
   // Create decoder.
   AudioDecoderOpus::Config dec_config;
   dec_config.num_channels = kNumChannels;
-  auto decoder = AudioDecoderOpus::MakeAudioDecoder(dec_config);
+  auto decoder = AudioDecoderOpus::MakeAudioDecoder(env, dec_config);
 
   // Open speech file.
   const std::string kInputFileName =
diff --git a/modules/audio_coding/neteq/audio_decoder_unittest.cc b/modules/audio_coding/neteq/audio_decoder_unittest.cc
index da2f496120..a39b46bb0b 100644
--- a/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -28,6 +28,7 @@
 #include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
 #include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
 #include "rtc_base/system/arch.h"
+#include "test/explicit_key_value_config.h"
 #include "test/gtest.h"
 #include "test/testsupport/file_utils.h"
 
@@ -35,6 +36,8 @@ namespace webrtc {
 
 namespace {
 
+using test::ExplicitKeyValueConfig;
+
 constexpr int kOverheadBytesPerPacket = 50;
 
 // The absolute difference between the input and output (the first channel) is
@@ -384,8 +387,8 @@ class AudioDecoderOpusTest
     codec_input_rate_hz_ = opus_sample_rate_hz_;
     frame_size_ = rtc::CheckedDivExact(opus_sample_rate_hz_, 100);
     data_length_ = 10 * frame_size_;
-    decoder_ =
-        new AudioDecoderOpusImpl(opus_num_channels_, opus_sample_rate_hz_);
+    decoder_ = new AudioDecoderOpusImpl(
+        ExplicitKeyValueConfig(""), opus_num_channels_, opus_sample_rate_hz_);
     AudioEncoderOpusConfig config;
     config.frame_size_ms = 10;
     config.sample_rate_hz = opus_sample_rate_hz_;
diff --git a/test/fuzzers/BUILD.gn b/test/fuzzers/BUILD.gn
index 708a4ebce9..1f0abe356a 100644
--- a/test/fuzzers/BUILD.gn
+++ b/test/fuzzers/BUILD.gn
@@ -330,6 +330,7 @@ webrtc_fuzzer_test("audio_decoder_opus_fuzzer") {
   sources = [ "audio_decoder_opus_fuzzer.cc" ]
   deps = [
     ":audio_decoder_fuzzer",
+    "..:explicit_key_value_config",
     "../../modules/audio_coding:webrtc_opus",
   ]
 }
@@ -338,6 +339,7 @@ webrtc_fuzzer_test("audio_decoder_opus_redundant_fuzzer") {
   sources = [ "audio_decoder_opus_redundant_fuzzer.cc" ]
   deps = [
     ":audio_decoder_fuzzer",
+    "..:explicit_key_value_config",
     "../../modules/audio_coding:webrtc_opus",
   ]
 }
diff --git a/test/fuzzers/audio_decoder_opus_fuzzer.cc b/test/fuzzers/audio_decoder_opus_fuzzer.cc
index a015f98b5b..84ee57f6f8 100644
--- a/test/fuzzers/audio_decoder_opus_fuzzer.cc
+++ b/test/fuzzers/audio_decoder_opus_fuzzer.cc
@@ -9,13 +9,15 @@
  */
 
 #include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
+#include "test/explicit_key_value_config.h"
 #include "test/fuzzers/audio_decoder_fuzzer.h"
 
 namespace webrtc {
 void FuzzOneInput(const uint8_t* data, size_t size) {
   const size_t channels = (size % 2) + 1;  // 1 or 2 channels.
-  AudioDecoderOpusImpl dec(channels);
-  const int kSampleRateHz = 48000;
+  const int kSampleRateHz = 48'000;
+  AudioDecoderOpusImpl dec(test::ExplicitKeyValueConfig(""), channels,
+                           kSampleRateHz);
   const size_t kAllocatedOuputSizeSamples = kSampleRateHz / 10;  // 100 ms.
   int16_t output[kAllocatedOuputSizeSamples];
   FuzzAudioDecoder(DecoderFunctionType::kNormalDecode, data, size, &dec,
diff --git a/test/fuzzers/audio_decoder_opus_redundant_fuzzer.cc b/test/fuzzers/audio_decoder_opus_redundant_fuzzer.cc
index efcba0f35d..3fec41e581 100644
--- a/test/fuzzers/audio_decoder_opus_redundant_fuzzer.cc
+++ b/test/fuzzers/audio_decoder_opus_redundant_fuzzer.cc
@@ -9,13 +9,15 @@
  */
 
 #include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
+#include "test/explicit_key_value_config.h"
 #include "test/fuzzers/audio_decoder_fuzzer.h"
 
 namespace webrtc {
 void FuzzOneInput(const uint8_t* data, size_t size) {
   const size_t channels = (size % 2) + 1;  // 1 or 2 channels.
-  AudioDecoderOpusImpl dec(channels);
-  const int kSampleRateHz = 48000;
+  const int kSampleRateHz = 48'000;
+  AudioDecoderOpusImpl dec(test::ExplicitKeyValueConfig(""), channels,
+                           kSampleRateHz);
   const size_t kAllocatedOuputSizeSamples = kSampleRateHz / 10;  // 100 ms.
   int16_t output[kAllocatedOuputSizeSamples];
   FuzzAudioDecoder(DecoderFunctionType::kRedundantDecode, data, size, &dec,