diff --git a/test/scenario/audio_stream.cc b/test/scenario/audio_stream.cc
index 521bcb7fbb..dc8e02ceda 100644
--- a/test/scenario/audio_stream.cc
+++ b/test/scenario/audio_stream.cc
@@ -9,6 +9,7 @@
  */
 #include "test/scenario/audio_stream.h"
 
+#include "rtc_base/bitrateallocationstrategy.h"
 #include "test/call_test.h"
 
 #if WEBRTC_ENABLE_PROTOBUF
@@ -131,8 +132,12 @@ SendAudioStream::SendAudioStream(
         {RtpExtension::kTransportSequenceNumberUri, 8}};
   }
 
-  if (config.stream.rate_allocation_priority) {
+  if (config.encoder.priority_rate) {
     send_config.track_id = sender->GetNextPriorityId();
+    sender_->call_->SetBitrateAllocationStrategy(
+        absl::make_unique<rtc::AudioPriorityBitrateAllocationStrategy>(
+            send_config.track_id,
+            config.encoder.priority_rate->bps<uint32_t>()));
   }
   send_stream_ = sender_->call_->CreateAudioSendStream(send_config);
   if (field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")) {
diff --git a/test/scenario/call_client.cc b/test/scenario/call_client.cc
index f51627b921..0fbd2bc8f2 100644
--- a/test/scenario/call_client.cc
+++ b/test/scenario/call_client.cc
@@ -147,12 +147,6 @@ CallClient::CallClient(Clock* clock,
                        fake_audio_setup_.audio_state)),
       transport_(clock_, call_.get()),
       header_parser_(RtpHeaderParser::Create()) {
-  if (!config.priority_target_rate.IsZero() &&
-      config.priority_target_rate.IsFinite()) {
-    call_->SetBitrateAllocationStrategy(
-        absl::make_unique<rtc::AudioPriorityBitrateAllocationStrategy>(
-            kPriorityStreamId, config.priority_target_rate.bps()));
-  }
 }  // namespace test
 
 CallClient::~CallClient() {
diff --git a/test/scenario/scenario_config.h b/test/scenario/scenario_config.h
index 8606d5f437..a4d55ab64f 100644
--- a/test/scenario/scenario_config.h
+++ b/test/scenario/scenario_config.h
@@ -51,7 +51,6 @@ struct TransportControllerConfig {
 
 struct CallClientConfig {
   TransportControllerConfig transport;
-  DataRate priority_target_rate = DataRate::Zero();
 };
 
 struct SimulatedTimeClientConfig {
@@ -155,6 +154,7 @@ struct AudioStreamConfig {
     absl::optional<DataRate> fixed_rate;
     absl::optional<DataRate> min_rate;
     absl::optional<DataRate> max_rate;
+    absl::optional<DataRate> priority_rate;
     TimeDelta initial_frame_length = TimeDelta::ms(20);
   } encoder;
   struct Stream {
@@ -162,7 +162,6 @@ struct AudioStreamConfig {
     Stream(const Stream&);
     ~Stream();
     bool in_bandwidth_estimation = false;
-    bool rate_allocation_priority = false;
   } stream;
   struct Render {
     std::string sync_group;