diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index aee2fc4305..08dd74591d 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -411,7 +411,7 @@ ChannelSend::ChannelSend(
 
   configuration.event_log = event_log_;
   configuration.rtt_stats = rtcp_rtt_stats;
-  if (!field_trials.IsEnabled("WebRTC-DisableRtxRateLimiter")) {
+  if (field_trials.IsDisabled("WebRTC-DisableRtxRateLimiter")) {
     configuration.retransmission_rate_limiter =
         retransmission_rate_limiter_.get();
   }
diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index 303a70b62a..1ace08fa32 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -226,7 +226,7 @@ std::vector<RtpStreamSender> CreateRtpStreamSenders(
   configuration.send_bitrate_observer = observers.bitrate_observer;
   configuration.send_packet_observer = observers.send_packet_observer;
   configuration.event_log = event_log;
-  if (!trials.IsEnabled("WebRTC-DisableRtxRateLimiter")) {
+  if (trials.IsDisabled("WebRTC-DisableRtxRateLimiter")) {
     configuration.retransmission_rate_limiter = retransmission_rate_limiter;
   }
   configuration.rtp_stats_callback = observers.rtp_stats;