diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 7c6d7940f6..aa2f6640c0 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -1638,14 +1638,6 @@ bool WebRtcVoiceMediaChannel::SetSendParameters(
     return false;
   }
 
-  if (params.max_bandwidth_bps >= 0) {
-    // Note that max_bandwidth_bps intentionally takes priority over the
-    // bitrate config for the codec.
-    bitrate_config_.max_bitrate_bps =
-        params.max_bandwidth_bps == 0 ? -1 : params.max_bandwidth_bps;
-  }
-  call_->SetBitrateConfig(bitrate_config_);
-
   if (!ValidateRtpExtensions(params.extensions)) {
     return false;
   }
@@ -1917,6 +1909,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
   // parameters.
   // TODO(solenberg): Refactor this logic once we create AudioEncoders here.
   webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec;
+  webrtc::Call::Config::BitrateConfig bitrate_config;
   {
     send_codec_spec.nack_enabled = send_codec_spec_.nack_enabled;
 
@@ -1930,7 +1923,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
 
     send_codec_spec.transport_cc_enabled = HasTransportCc(*codec);
     send_codec_spec.nack_enabled = HasNack(*codec);
-    bitrate_config_ = GetBitrateConfigForCodec(*codec);
+    bitrate_config = GetBitrateConfigForCodec(*codec);
 
     // For Opus as the send codec, we are to determine inband FEC, maximum
     // playback rate, and opus internal dtx.
@@ -2007,8 +2000,9 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
   } else {
     // If the codec isn't changing, set the start bitrate to -1 which means
     // "unchanged" so that BWE isn't affected.
-    bitrate_config_.start_bitrate_bps = -1;
+    bitrate_config.start_bitrate_bps = -1;
   }
+  call_->SetBitrateConfig(bitrate_config);
 
   // Check if the transport cc feedback or NACK status has changed on the
   // preferred send codec, and in that case reconfigure all receive streams.
diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h
index c777c6c03e..851ca2df8e 100644
--- a/webrtc/media/engine/webrtcvoiceengine.h
+++ b/webrtc/media/engine/webrtcvoiceengine.h
@@ -262,7 +262,6 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
   bool playout_ = false;
   bool send_ = false;
   webrtc::Call* const call_ = nullptr;
-  webrtc::Call::Config::BitrateConfig bitrate_config_;
 
   // Queue of unsignaled SSRCs; oldest at the beginning.
   std::vector<uint32_t> unsignaled_recv_ssrcs_;
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index 2d41ecd43b..51d20874a8 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -1533,16 +1533,16 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCapsMinAndStartBitrate) {
 }
 
 TEST_F(WebRtcVoiceEngineTestFake,
-       SetMaxSendBandwidthShouldPreserveOtherBitrates) {
+       SetMaxSendBandwidthForAudioDoesntAffectBwe) {
   SetSendCodecsShouldWorkForBitrates("100", 100000, "150", 150000, "200",
                                      200000);
-  send_parameters_.max_bandwidth_bps = 300000;
+  send_parameters_.max_bandwidth_bps = 100000;
   SetSendParameters(send_parameters_);
   EXPECT_EQ(100000, call_.GetConfig().bitrate_config.min_bitrate_bps)
       << "Setting max bitrate should keep previous min bitrate.";
   EXPECT_EQ(-1, call_.GetConfig().bitrate_config.start_bitrate_bps)
       << "Setting max bitrate should not reset start bitrate.";
-  EXPECT_EQ(300000, call_.GetConfig().bitrate_config.max_bitrate_bps);
+  EXPECT_EQ(200000, call_.GetConfig().bitrate_config.max_bitrate_bps);
 }
 
 // Test that we can enable NACK with opus as caller.