diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index 7e88b59d83..b7616d1a1f 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -498,7 +498,7 @@ void RtpVideoSender::SetActiveModules(const std::vector<bool>& active_modules) {
 void RtpVideoSender::SetActiveModulesLocked(
     const std::vector<bool>& active_modules) {
   RTC_DCHECK_RUN_ON(&transport_checker_);
-  RTC_DCHECK_EQ(rtp_streams_.size(), active_modules.size());
+  RTC_CHECK_EQ(rtp_streams_.size(), active_modules.size());
   active_ = false;
   for (size_t i = 0; i < active_modules.size(); ++i) {
     if (active_modules[i]) {
diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index af5d0c7fe9..35a5eb479d 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -2166,14 +2166,9 @@ void WebRtcVideoSendChannel::WebRtcVideoSendStream::ReconfigureEncoder(
   // layers specified by `scalability_mode`), the number of streams can change.
   bool num_streams_changed = parameters_.encoder_config.number_of_streams !=
                              encoder_config.number_of_streams;
-  bool scalability_mode_used = !codec_settings.codec.scalability_modes.empty();
-  bool scalability_modes = absl::c_any_of(
-      rtp_parameters_.encodings,
-      [](const auto& e) { return e.scalability_mode.has_value(); });
-
   parameters_.encoder_config = std::move(encoder_config);
 
-  if (num_streams_changed && (scalability_mode_used != scalability_modes)) {
+  if (num_streams_changed) {
     // The app is switching between legacy and standard modes, recreate instead
     // of reconfiguring to avoid number of streams not matching in lower layers.
     RecreateWebRtcStream();