diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index fc6caf5105..a72292dfae 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -804,7 +804,9 @@ void AudioSendStream::ConfigureBitrateObserver() {
       MediaStreamAllocationConfig{
           constraints.min.bps<uint32_t>(), constraints.max.bps<uint32_t>(), 0,
           allocation_settings_.DefaultPriorityBitrate().bps(), true,
-          config_.track_id, config_.bitrate_priority});
+          config_.track_id,
+          allocation_settings_.BitratePriority().value_or(
+              config_.bitrate_priority)});
 }
 
 void AudioSendStream::RemoveBitrateObserver() {
diff --git a/call/bitrate_allocator.cc b/call/bitrate_allocator.cc
index 9d2705f66c..7d5b52da90 100644
--- a/call/bitrate_allocator.cc
+++ b/call/bitrate_allocator.cc
@@ -67,7 +67,9 @@ BitrateAllocator::BitrateAllocator(Clock* clock, LimitObserver* limit_observer)
       total_requested_max_bitrate_(0),
       bitrate_allocation_strategy_(nullptr),
       transmission_max_bitrate_multiplier_(
-          GetTransmissionMaxBitrateMultiplier()) {
+          GetTransmissionMaxBitrateMultiplier()),
+      ignore_injected_strategy_(
+          field_trial::IsEnabled("WebRTC-IgnoreInjectedAllocationStrategy")) {
   sequenced_checker_.Detach();
 }
 
@@ -315,7 +317,7 @@ BitrateAllocator::ObserverAllocation BitrateAllocator::AllocateBitrates(
   if (bitrate_observer_configs_.empty())
     return ObserverAllocation();
 
-  if (bitrate_allocation_strategy_ != nullptr) {
+  if (!ignore_injected_strategy_ && bitrate_allocation_strategy_ != nullptr) {
     // Note: This intentionally causes slicing, we only copy the fields in
     // ObserverConfig that are inherited from TrackConfig.
     std::vector<rtc::BitrateAllocationStrategy::TrackConfig> track_configs(
diff --git a/call/bitrate_allocator.h b/call/bitrate_allocator.h
index f35f3f7d7d..8611a8967b 100644
--- a/call/bitrate_allocator.h
+++ b/call/bitrate_allocator.h
@@ -246,6 +246,7 @@ class BitrateAllocator : public BitrateAllocatorInterface {
   std::unique_ptr<rtc::BitrateAllocationStrategy> bitrate_allocation_strategy_
       RTC_GUARDED_BY(&sequenced_checker_);
   const uint8_t transmission_max_bitrate_multiplier_;
+  const bool ignore_injected_strategy_;
 };
 
 }  // namespace webrtc
diff --git a/rtc_base/experiments/audio_allocation_settings.cc b/rtc_base/experiments/audio_allocation_settings.cc
index 6579531da6..66d9d4785b 100644
--- a/rtc_base/experiments/audio_allocation_settings.cc
+++ b/rtc_base/experiments/audio_allocation_settings.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include "rtc_base/experiments/audio_allocation_settings.h"
+
 #include "system_wrappers/include/field_trial.h"
 
 namespace webrtc {
@@ -22,7 +23,8 @@ AudioAllocationSettings::AudioAllocationSettings()
       send_side_bwe_with_overhead_("Enabled"),
       min_bitrate_("min"),
       max_bitrate_("max"),
-      priority_bitrate_("prio", DataRate::Zero()) {
+      priority_bitrate_("prio_rate", DataRate::Zero()),
+      bitrate_priority_("rate_prio") {
   ParseFieldTrial({&audio_send_side_bwe_},
                   field_trial::FindFullName("WebRTC-Audio-SendSideBwe"));
   ParseFieldTrial({&allocate_audio_without_feedback_},
@@ -32,8 +34,9 @@ AudioAllocationSettings::AudioAllocationSettings()
 
   ParseFieldTrial({&send_side_bwe_with_overhead_},
                   field_trial::FindFullName("WebRTC-SendSideBwe-WithOverhead"));
-  ParseFieldTrial({&min_bitrate_, &max_bitrate_, &priority_bitrate_},
-                  field_trial::FindFullName("WebRTC-Audio-Allocation"));
+  ParseFieldTrial(
+      {&min_bitrate_, &max_bitrate_, &priority_bitrate_, &bitrate_priority_},
+      field_trial::FindFullName("WebRTC-Audio-Allocation"));
 
   // TODO(mflodman): Keep testing this and set proper values.
   // Note: This is an early experiment currently only supported by Opus.
@@ -114,5 +117,8 @@ DataRate AudioAllocationSettings::DefaultPriorityBitrate() const {
   }
   return priority_bitrate_.Get() + max_overhead;
 }
+absl::optional<double> AudioAllocationSettings::BitratePriority() const {
+  return bitrate_priority_.GetOptional();
+}
 
 }  // namespace webrtc
diff --git a/rtc_base/experiments/audio_allocation_settings.h b/rtc_base/experiments/audio_allocation_settings.h
index c0999735ee..d85b00766a 100644
--- a/rtc_base/experiments/audio_allocation_settings.h
+++ b/rtc_base/experiments/audio_allocation_settings.h
@@ -72,6 +72,10 @@ class AudioAllocationSettings {
   // divide bitrate evently between audio and video above this bitrate.
   DataRate DefaultPriorityBitrate() const;
 
+  // The bitrate priority is used to determine how much of the available bitrate
+  // beyond the min or priority bitrate audio streams should receive.
+  absl::optional<double> BitratePriority() const;
+
  private:
   FieldTrialFlag audio_send_side_bwe_;
   FieldTrialFlag allocate_audio_without_feedback_;
@@ -83,6 +87,7 @@ class AudioAllocationSettings {
   FieldTrialOptional<DataRate> min_bitrate_;
   FieldTrialOptional<DataRate> max_bitrate_;
   FieldTrialParameter<DataRate> priority_bitrate_;
+  FieldTrialOptional<double> bitrate_priority_;
 };
 }  // namespace webrtc