diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index 467647b..cdbab66 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -18,6 +18,7 @@
 #include "api/audio_codecs/audio_format.h"
 #include "api/call/audio_sink.h"
 #include "api/rtp_parameters.h"
+#include "api/sequence_checker.h"
 #include "audio/audio_send_stream.h"
 #include "audio/audio_state.h"
 #include "audio/channel_receive.h"
@@ -118,7 +119,8 @@
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
     webrtc::RtcEventLog* event_log,
     std::unique_ptr<voe::ChannelReceiveInterface> channel_receive)
-    : audio_state_(audio_state),
+    : config_(config),
+      audio_state_(audio_state),
       source_tracker_(clock),
       channel_receive_(std::move(channel_receive)) {
   RTC_LOG(LS_INFO) << "AudioReceiveStream: " << config.rtp.remote_ssrc;
@@ -140,7 +142,15 @@
   // Register with transport.
   rtp_stream_receiver_ = receiver_controller->CreateReceiver(
       config.rtp.remote_ssrc, channel_receive_.get());
-  ConfigureStream(this, config, true);
+
+  // Complete configuration.
+  // TODO(solenberg): Config NACK history window (which is a packet count),
+  // using the actual packet size for the configured codec.
+  channel_receive_->SetNACKStatus(config.rtp.nack.rtp_history_ms != 0,
+                                  config.rtp.nack.rtp_history_ms / 20);
+  channel_receive_->SetReceiveCodecs(config.decoder_map);
+  channel_receive_->SetDepacketizerToDecoderFrameTransformer(
+      config.frame_transformer);
 }
 
 AudioReceiveStream::~AudioReceiveStream() {
@@ -154,7 +164,34 @@
 void AudioReceiveStream::Reconfigure(
     const webrtc::AudioReceiveStream::Config& config) {
   RTC_DCHECK(worker_thread_checker_.IsCurrent());
-  ConfigureStream(this, config, false);
+
+  // Configuration parameters which cannot be changed.
+  RTC_DCHECK(config_.rtp.remote_ssrc == config.rtp.remote_ssrc);
+  RTC_DCHECK(config_.rtcp_send_transport == config.rtcp_send_transport);
+  // Decoder factory cannot be changed because it is configured at
+  // voe::Channel construction time.
+  RTC_DCHECK(config_.decoder_factory == config.decoder_factory);
+
+  // SSRC can't be changed mid-stream.
+  RTC_DCHECK_EQ(config_.rtp.local_ssrc, config.rtp.local_ssrc);
+  RTC_DCHECK_EQ(config_.rtp.remote_ssrc, config.rtp.remote_ssrc);
+
+  // TODO(solenberg): Config NACK history window (which is a packet count),
+  // using the actual packet size for the configured codec.
+  if (config_.rtp.nack.rtp_history_ms != config.rtp.nack.rtp_history_ms) {
+    channel_receive_->SetNACKStatus(config.rtp.nack.rtp_history_ms != 0,
+                                    config.rtp.nack.rtp_history_ms / 20);
+  }
+  if (config_.decoder_map != config.decoder_map) {
+    channel_receive_->SetReceiveCodecs(config.decoder_map);
+  }
+
+  if (config_.frame_transformer != config.frame_transformer) {
+    channel_receive_->SetDepacketizerToDecoderFrameTransformer(
+        config.frame_transformer);
+  }
+
+  config_ = config;
 }
 
 void AudioReceiveStream::Start() {
@@ -384,50 +421,5 @@
   RTC_DCHECK(audio_state);
   return audio_state;
 }
-
-void AudioReceiveStream::ConfigureStream(AudioReceiveStream* stream,
-                                         const Config& new_config,
-                                         bool first_time) {
-  RTC_LOG(LS_INFO) << "AudioReceiveStream::ConfigureStream: "
-                   << new_config.ToString();
-  RTC_DCHECK(stream);
-  const auto& channel_receive = stream->channel_receive_;
-  const auto& old_config = stream->config_;
-
-  // Configuration parameters which cannot be changed.
-  RTC_DCHECK(first_time ||
-             old_config.rtp.remote_ssrc == new_config.rtp.remote_ssrc);
-  RTC_DCHECK(first_time ||
-             old_config.rtcp_send_transport == new_config.rtcp_send_transport);
-  // Decoder factory cannot be changed because it is configured at
-  // voe::Channel construction time.
-  RTC_DCHECK(first_time ||
-             old_config.decoder_factory == new_config.decoder_factory);
-
-  if (!first_time) {
-    // SSRC can't be changed mid-stream.
-    RTC_DCHECK_EQ(old_config.rtp.local_ssrc, new_config.rtp.local_ssrc);
-    RTC_DCHECK_EQ(old_config.rtp.remote_ssrc, new_config.rtp.remote_ssrc);
-  }
-
-  // TODO(solenberg): Config NACK history window (which is a packet count),
-  // using the actual packet size for the configured codec.
-  if (first_time || old_config.rtp.nack.rtp_history_ms !=
-                        new_config.rtp.nack.rtp_history_ms) {
-    channel_receive->SetNACKStatus(new_config.rtp.nack.rtp_history_ms != 0,
-                                   new_config.rtp.nack.rtp_history_ms / 20);
-  }
-  if (first_time || old_config.decoder_map != new_config.decoder_map) {
-    channel_receive->SetReceiveCodecs(new_config.decoder_map);
-  }
-
-  if (first_time ||
-      old_config.frame_transformer != new_config.frame_transformer) {
-    channel_receive->SetDepacketizerToDecoderFrameTransformer(
-        new_config.frame_transformer);
-  }
-
-  stream->config_ = new_config;
-}
 }  // namespace internal
 }  // namespace webrtc
diff --git a/audio/audio_receive_stream.h b/audio/audio_receive_stream.h
index a8438c2..769578b 100644
--- a/audio/audio_receive_stream.h
+++ b/audio/audio_receive_stream.h
@@ -102,10 +102,6 @@
   const AudioSendStream* GetAssociatedSendStreamForTesting() const;
 
  private:
-  static void ConfigureStream(AudioReceiveStream* stream,
-                              const Config& new_config,
-                              bool first_time);
-
   AudioState* audio_state() const;
 
   SequenceChecker worker_thread_checker_;