diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index 8c073d9c8c..f64d3051e6 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -94,10 +94,7 @@ AudioReceiveStream::AudioReceiveStream(
                channel_proxy_->GetAudioDecoderFactory());
 
   channel_proxy_->RegisterExternalTransport(config.rtcp_send_transport);
-
-  for (const auto& kv : config.decoder_map) {
-    channel_proxy_->SetRecPayloadType(kv.first, kv.second);
-  }
+  channel_proxy_->SetReceiveCodecs(config.decoder_map);
 
   for (const auto& extension : config.rtp.extensions) {
     if (extension.uri == RtpExtension::kAudioLevelUri) {
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index e6b81a8bab..9025550b4f 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <map>
 #include <string>
 #include <vector>
 
@@ -111,6 +112,11 @@ struct ConfigHelper {
               .Times(1)
               .After(expect_set);
           EXPECT_CALL(*channel_proxy_, DisassociateSendChannel()).Times(1);
+          EXPECT_CALL(*channel_proxy_, SetReceiveCodecs(_))
+              .WillRepeatedly(
+                  Invoke([](const std::map<int, SdpAudioFormat>& codecs) {
+                    EXPECT_THAT(codecs, testing::IsEmpty());
+                  }));
           return channel_proxy_;
         }));
     stream_config_.voe_channel_id = kChannelId;
diff --git a/webrtc/call/call_unittest.cc b/webrtc/call/call_unittest.cc
index 99d6812d68..a45b91cb72 100644
--- a/webrtc/call/call_unittest.cc
+++ b/webrtc/call/call_unittest.cc
@@ -9,6 +9,7 @@
  */
 
 #include <list>
+#include <map>
 #include <memory>
 
 #include "webrtc/call/audio_state.h"
@@ -141,6 +142,11 @@ TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_RecvFirst) {
             new testing::NiceMock<test::MockVoEChannelProxy>();
         EXPECT_CALL(*channel_proxy, GetAudioDecoderFactory())
             .WillRepeatedly(testing::ReturnRef(decoder_factory));
+        EXPECT_CALL(*channel_proxy, SetReceiveCodecs(testing::_))
+            .WillRepeatedly(testing::Invoke(
+                [](const std::map<int, SdpAudioFormat>& codecs) {
+                  EXPECT_THAT(codecs, testing::IsEmpty());
+                }));
         // If being called for the send channel, save a pointer to the channel
         // proxy for later.
         if (channel_id == kRecvChannelId) {
@@ -188,6 +194,11 @@ TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_SendFirst) {
             new testing::NiceMock<test::MockVoEChannelProxy>();
         EXPECT_CALL(*channel_proxy, GetAudioDecoderFactory())
             .WillRepeatedly(testing::ReturnRef(decoder_factory));
+        EXPECT_CALL(*channel_proxy, SetReceiveCodecs(testing::_))
+            .WillRepeatedly(testing::Invoke(
+                [](const std::map<int, SdpAudioFormat>& codecs) {
+                  EXPECT_THAT(codecs, testing::IsEmpty());
+                }));
         // If being called for the send channel, save a pointer to the channel
         // proxy for later.
         if (channel_id == kRecvChannelId) {
diff --git a/webrtc/media/engine/fakewebrtcvoiceengine.h b/webrtc/media/engine/fakewebrtcvoiceengine.h
index e7d95d09e7..180d06bcaf 100644
--- a/webrtc/media/engine/fakewebrtcvoiceengine.h
+++ b/webrtc/media/engine/fakewebrtcvoiceengine.h
@@ -117,8 +117,6 @@ class FakeWebRtcVoiceEngine
       return -1;
     }
     Channel* ch = new Channel();
-    auto db = webrtc::acm2::RentACodec::Database();
-    ch->recv_codecs.assign(db.begin(), db.end());
     ch->neteq_capacity = config.acm_config.neteq_config.max_packets_in_buffer;
     ch->neteq_fast_accelerate =
         config.acm_config.neteq_config.enable_fast_accelerate;
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 7c6d7940f6..31a9d27184 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -132,13 +132,6 @@ std::string ToString(const AudioCodec& codec) {
   return ss.str();
 }
 
-std::string ToString(const webrtc::CodecInst& codec) {
-  std::stringstream ss;
-  ss << codec.plname << "/" << codec.plfreq << "/" << codec.channels
-     << " (" << codec.pltype << ")";
-  return ss.str();
-}
-
 bool IsCodec(const AudioCodec& codec, const char* ref_name) {
   return (_stricmp(codec.name.c_str(), ref_name) == 0);
 }
@@ -1466,7 +1459,8 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
       const std::vector<webrtc::RtpExtension>& extensions,
       webrtc::Call* call,
       webrtc::Transport* rtcp_send_transport,
-      const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory)
+      const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
+      const std::map<int, webrtc::SdpAudioFormat>& decoder_map)
       : call_(call), config_() {
     RTC_DCHECK_GE(ch, 0);
     RTC_DCHECK(call);
@@ -1479,6 +1473,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
     config_.voe_channel_id = ch;
     config_.sync_group = sync_group;
     config_.decoder_factory = decoder_factory;
+    config_.decoder_map = decoder_map;
     RecreateAudioReceiveStream();
   }
 
@@ -1867,8 +1862,9 @@ bool WebRtcVoiceMediaChannel::SetRecvCodecs(
     ChangePlayout(false);
   }
 
+  decoder_map_ = std::move(decoder_map);
   for (auto& kv : recv_streams_) {
-    kv.second->RecreateAudioReceiveStream(decoder_map);
+    kv.second->RecreateAudioReceiveStream(decoder_map_);
   }
   recv_codecs_ = codecs;
 
@@ -2225,38 +2221,12 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
     return false;
   }
 
-  // Turn off all supported codecs.
-  // TODO(solenberg): Remove once "no codecs" is the default state of a stream.
-  for (webrtc::CodecInst voe_codec : webrtc::acm2::RentACodec::Database()) {
-    voe_codec.pltype = -1;
-    if (engine()->voe()->codec()->SetRecPayloadType(channel, voe_codec) == -1) {
-      LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec));
-      DeleteVoEChannel(channel);
-      return false;
-    }
-  }
-
-  // Only enable those configured for this channel.
-  for (const auto& codec : recv_codecs_) {
-    webrtc::CodecInst voe_codec = {0};
-    if (WebRtcVoiceEngine::ToCodecInst(codec, &voe_codec)) {
-      voe_codec.pltype = codec.id;
-      if (engine()->voe()->codec()->SetRecPayloadType(
-          channel, voe_codec) == -1) {
-        LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec));
-        DeleteVoEChannel(channel);
-        return false;
-      }
-    }
-  }
-
   recv_streams_.insert(std::make_pair(
-      ssrc, new WebRtcAudioReceiveStream(channel, ssrc, receiver_reports_ssrc_,
-                                         recv_transport_cc_enabled_,
-                                         recv_nack_enabled_,
-                                         sp.sync_label, recv_rtp_extensions_,
-                                         call_, this,
-                                         engine()->decoder_factory_)));
+      ssrc,
+      new WebRtcAudioReceiveStream(
+          channel, ssrc, receiver_reports_ssrc_, recv_transport_cc_enabled_,
+          recv_nack_enabled_, sp.sync_label, recv_rtp_extensions_, call_, this,
+          engine()->decoder_factory_, decoder_map_)));
   recv_streams_[ssrc]->SetPlayout(playout_);
 
   return true;
diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h
index c777c6c03e..64f7afabf6 100644
--- a/webrtc/media/engine/webrtcvoiceengine.h
+++ b/webrtc/media/engine/webrtcvoiceengine.h
@@ -251,7 +251,12 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
 
   WebRtcVoiceEngine* const engine_ = nullptr;
   std::vector<AudioCodec> send_codecs_;
+
+  // TODO(kwiberg): decoder_map_ and recv_codecs_ store the exact same
+  // information, in slightly different formats. Eliminate recv_codecs_.
+  std::map<int, webrtc::SdpAudioFormat> decoder_map_;
   std::vector<AudioCodec> recv_codecs_;
+
   int max_send_bitrate_bps_ = 0;
   AudioOptions options_;
   rtc::Optional<int> dtmf_payload_type_;
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index 2d41ecd43b..3777b3daa3 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -31,6 +31,7 @@
 #include "webrtc/test/gtest.h"
 #include "webrtc/voice_engine/transmit_mixer.h"
 
+using testing::ContainerEq;
 using testing::Return;
 using testing::StrictMock;
 
@@ -795,26 +796,12 @@ TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecs) {
   parameters.codecs[2].id = 126;
   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
   EXPECT_TRUE(AddRecvStream(kSsrcX));
-  int channel_num = voe_.GetLastChannel();
-
-  webrtc::CodecInst gcodec;
-  rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "ISAC");
-  gcodec.plfreq = 16000;
-  gcodec.channels = 1;
-  EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num, gcodec));
-  EXPECT_EQ(106, gcodec.pltype);
-  EXPECT_STREQ("ISAC", gcodec.plname);
-
-  rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "telephone-event");
-  gcodec.plfreq = 8000;
-  EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num, gcodec));
-  EXPECT_EQ(126, gcodec.pltype);
-  EXPECT_STREQ("telephone-event", gcodec.plname);
-
-  gcodec.plfreq = 32000;
-  EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num, gcodec));
-  EXPECT_EQ(107, gcodec.pltype);
-  EXPECT_STREQ("telephone-event", gcodec.plname);
+  EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
+              (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
+                  {{0, {"PCMU", 8000, 1}},
+                   {106, {"ISAC", 16000, 1}},
+                   {126, {"telephone-event", 8000, 1}},
+                   {107, {"telephone-event", 32000, 1}}})));
 }
 
 // Test that we fail to set an unknown inbound codec.
@@ -845,16 +832,11 @@ TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpusNoStereo) {
   parameters.codecs.push_back(kOpusCodec);
   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
   EXPECT_TRUE(AddRecvStream(kSsrcX));
-  int channel_num = voe_.GetLastChannel();
-  webrtc::CodecInst opus;
-  cricket::WebRtcVoiceEngine::ToCodecInst(kOpusCodec, &opus);
-  // Even without stereo parameters, recv codecs still specify channels = 2.
-  EXPECT_EQ(2, opus.channels);
-  EXPECT_EQ(111, opus.pltype);
-  EXPECT_STREQ("opus", opus.plname);
-  opus.pltype = 0;
-  EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num, opus));
-  EXPECT_EQ(111, opus.pltype);
+  EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
+              (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
+                  {{0, {"PCMU", 8000, 1}},
+                   {103, {"ISAC", 16000, 1}},
+                   {111, {"opus", 48000, 2}}})));
 }
 
 // Test that we can decode OPUS with stereo = 0.
@@ -867,16 +849,11 @@ TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus0Stereo) {
   parameters.codecs[2].params["stereo"] = "0";
   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
   EXPECT_TRUE(AddRecvStream(kSsrcX));
-  int channel_num2 = voe_.GetLastChannel();
-  webrtc::CodecInst opus;
-  cricket::WebRtcVoiceEngine::ToCodecInst(kOpusCodec, &opus);
-  // Even when stereo is off, recv codecs still specify channels = 2.
-  EXPECT_EQ(2, opus.channels);
-  EXPECT_EQ(111, opus.pltype);
-  EXPECT_STREQ("opus", opus.plname);
-  opus.pltype = 0;
-  EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, opus));
-  EXPECT_EQ(111, opus.pltype);
+  EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
+              (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
+                  {{0, {"PCMU", 8000, 1}},
+                   {103, {"ISAC", 16000, 1}},
+                   {111, {"opus", 48000, 2, {{"stereo", "0"}}}}})));
 }
 
 // Test that we can decode OPUS with stereo = 1.
@@ -889,15 +866,11 @@ TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus1Stereo) {
   parameters.codecs[2].params["stereo"] = "1";
   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
   EXPECT_TRUE(AddRecvStream(kSsrcX));
-  int channel_num2 = voe_.GetLastChannel();
-  webrtc::CodecInst opus;
-  cricket::WebRtcVoiceEngine::ToCodecInst(kOpusCodec, &opus);
-  EXPECT_EQ(2, opus.channels);
-  EXPECT_EQ(111, opus.pltype);
-  EXPECT_STREQ("opus", opus.plname);
-  opus.pltype = 0;
-  EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, opus));
-  EXPECT_EQ(111, opus.pltype);
+  EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
+              (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
+                  {{0, {"PCMU", 8000, 1}},
+                   {103, {"ISAC", 16000, 1}},
+                   {111, {"opus", 48000, 2, {{"stereo", "1"}}}}})));
 }
 
 // Test that changes to recv codecs are applied to all streams.
@@ -911,28 +884,15 @@ TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithMultipleStreams) {
   parameters.codecs[0].id = 106;  // collide with existing CN 32k
   parameters.codecs[2].id = 126;
   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
-  EXPECT_TRUE(AddRecvStream(kSsrcX));
-  int channel_num2 = voe_.GetLastChannel();
-
-  webrtc::CodecInst gcodec;
-  rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "ISAC");
-  gcodec.plfreq = 16000;
-  gcodec.channels = 1;
-  EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, gcodec));
-  EXPECT_EQ(106, gcodec.pltype);
-  EXPECT_STREQ("ISAC", gcodec.plname);
-
-  rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "telephone-event");
-  gcodec.plfreq = 8000;
-  gcodec.channels = 1;
-  EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, gcodec));
-  EXPECT_EQ(126, gcodec.pltype);
-  EXPECT_STREQ("telephone-event", gcodec.plname);
-
-  gcodec.plfreq = 32000;
-  EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, gcodec));
-  EXPECT_EQ(107, gcodec.pltype);
-  EXPECT_STREQ("telephone-event", gcodec.plname);
+  for (const auto& ssrc : {kSsrcX, kSsrcY}) {
+    EXPECT_TRUE(AddRecvStream(ssrc));
+    EXPECT_THAT(GetRecvStreamConfig(ssrc).decoder_map,
+                (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
+                    {{0, {"PCMU", 8000, 1}},
+                     {106, {"ISAC", 16000, 1}},
+                     {126, {"telephone-event", 8000, 1}},
+                     {107, {"telephone-event", 32000, 1}}})));
+  }
 }
 
 TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsAfterAddingStreams) {
@@ -2961,12 +2921,9 @@ TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamUnsupportedCodec) {
   parameters.codecs.push_back(kPcmuCodec);
   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
   EXPECT_TRUE(AddRecvStream(kSsrcX));
-  int channel_num2 = voe_.GetLastChannel();
-  webrtc::CodecInst gcodec;
-  rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "opus");
-  gcodec.plfreq = 48000;
-  gcodec.channels = 2;
-  EXPECT_EQ(-1, voe_.GetRecPayloadType(channel_num2, gcodec));
+  EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
+              (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
+                  {{0, {"PCMU", 8000, 1}}, {103, {"ISAC", 16000, 1}}})));
 }
 
 // Test that we properly clean up any streams that were added, even if
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
index 21dbc747ec..4980d279e7 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
@@ -179,6 +179,10 @@ int AcmReceiver::GetAudio(int desired_freq_hz,
   return 0;
 }
 
+void AcmReceiver::SetCodecs(const std::map<int, SdpAudioFormat>& codecs) {
+  neteq_->SetCodecs(codecs);
+}
+
 int32_t AcmReceiver::AddCodec(int acm_codec_id,
                               uint8_t payload_type,
                               size_t channels,
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.h b/webrtc/modules/audio_coding/acm2/acm_receiver.h
index 63ed43d8fa..8f0963f0af 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.h
@@ -79,6 +79,9 @@ class AcmReceiver {
   //
   int GetAudio(int desired_freq_hz, AudioFrame* audio_frame, bool* muted);
 
+  // Replace the current set of decoders with the specified set.
+  void SetCodecs(const std::map<int, SdpAudioFormat>& codecs);
+
   //
   // Adds a new codec to the NetEq codec database.
   //
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
index daeea3577e..bc814f6d6f 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
@@ -121,6 +121,8 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
   // Get current playout frequency.
   int PlayoutFrequency() const override;
 
+  void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
+
   bool RegisterReceiveCodec(int rtp_payload_type,
                             const SdpAudioFormat& audio_format) override;
 
@@ -318,16 +320,6 @@ void UpdateCodecTypeHistogram(size_t codec_type) {
           webrtc::AudioEncoder::CodecType::kMaxLoggedAudioCodecTypes));
 }
 
-// TODO(turajs): the same functionality is used in NetEq. If both classes
-// need them, make it a static function in ACMCodecDB.
-bool IsCodecRED(const CodecInst& codec) {
-  return (STR_CASE_CMP(codec.plname, "RED") == 0);
-}
-
-bool IsCodecCN(const CodecInst& codec) {
-  return (STR_CASE_CMP(codec.plname, "CN") == 0);
-}
-
 // Stereo-to-mono can be used as in-place.
 int DownMix(const AudioFrame& frame,
             size_t length_out_buff,
@@ -956,19 +948,6 @@ int AudioCodingModuleImpl::InitializeReceiverSafe() {
   receiver_.SetMaximumDelay(0);
   receiver_.FlushBuffers();
 
-  // Register RED and CN.
-  auto db = acm2::RentACodec::Database();
-  for (size_t i = 0; i < db.size(); i++) {
-    if (IsCodecRED(db[i]) || IsCodecCN(db[i])) {
-      if (receiver_.AddCodec(static_cast<int>(i),
-                             static_cast<uint8_t>(db[i].pltype), 1,
-                             db[i].plfreq, nullptr, db[i].plname) < 0) {
-        WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
-                     "Cannot register master codec.");
-        return -1;
-      }
-    }
-  }
   receiver_initialized_ = true;
   return 0;
 }
@@ -987,6 +966,12 @@ int AudioCodingModuleImpl::PlayoutFrequency() const {
   return receiver_.last_output_sample_rate_hz();
 }
 
+void AudioCodingModuleImpl::SetReceiveCodecs(
+    const std::map<int, SdpAudioFormat>& codecs) {
+  rtc::CritScope lock(&acm_crit_sect_);
+  receiver_.SetCodecs(codecs);
+}
+
 bool AudioCodingModuleImpl::RegisterReceiveCodec(
     int rtp_payload_type,
     const SdpAudioFormat& audio_format) {
diff --git a/webrtc/modules/audio_coding/include/audio_coding_module.h b/webrtc/modules/audio_coding/include/audio_coding_module.h
index 1893b626de..b5cbff2d3a 100644
--- a/webrtc/modules/audio_coding/include/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/include/audio_coding_module.h
@@ -485,6 +485,10 @@ class AudioCodingModule {
   //
   virtual int32_t PlayoutFrequency() const = 0;
 
+  // Replace any existing decoders with the given payload type -> decoder map.
+  virtual void SetReceiveCodecs(
+      const std::map<int, SdpAudioFormat>& codecs) = 0;
+
   // Registers a decoder for the given payload type. Returns true iff
   // successful.
   virtual bool RegisterReceiveCodec(int rtp_payload_type,
diff --git a/webrtc/modules/audio_coding/neteq/decoder_database.cc b/webrtc/modules/audio_coding/neteq/decoder_database.cc
index d147d676ed..8cd438656a 100644
--- a/webrtc/modules/audio_coding/neteq/decoder_database.cc
+++ b/webrtc/modules/audio_coding/neteq/decoder_database.cc
@@ -123,6 +123,38 @@ void DecoderDatabase::Reset() {
   active_cng_decoder_type_ = -1;
 }
 
+std::vector<int> DecoderDatabase::SetCodecs(
+    const std::map<int, SdpAudioFormat>& codecs) {
+  // First collect all payload types that we'll remove or reassign, then remove
+  // them from the database.
+  std::vector<int> changed_payload_types;
+  for (const std::pair<uint8_t, const DecoderInfo&> kv : decoders_) {
+    auto i = codecs.find(kv.first);
+    if (i == codecs.end() || i->second != kv.second.GetFormat()) {
+      changed_payload_types.push_back(kv.first);
+    }
+  }
+  for (int pl_type : changed_payload_types) {
+    Remove(pl_type);
+  }
+
+  // Enter the new and changed payload type mappings into the database.
+  for (const auto& kv : codecs) {
+    const int& rtp_payload_type = kv.first;
+    const SdpAudioFormat& audio_format = kv.second;
+    RTC_DCHECK_GE(rtp_payload_type, 0);
+    RTC_DCHECK_LE(rtp_payload_type, 0x7f);
+    if (decoders_.count(rtp_payload_type) == 0) {
+      decoders_.insert(std::make_pair(
+          rtp_payload_type, DecoderInfo(audio_format, decoder_factory_.get())));
+    } else {
+      // The mapping for this payload type hasn't changed.
+    }
+  }
+
+  return changed_payload_types;
+}
+
 int DecoderDatabase::RegisterPayload(uint8_t rtp_payload_type,
                                      NetEqDecoder codec_type,
                                      const std::string& name) {
diff --git a/webrtc/modules/audio_coding/neteq/decoder_database.h b/webrtc/modules/audio_coding/neteq/decoder_database.h
index ec470f8d5f..35d1d0da08 100644
--- a/webrtc/modules/audio_coding/neteq/decoder_database.h
+++ b/webrtc/modules/audio_coding/neteq/decoder_database.h
@@ -149,6 +149,11 @@ class DecoderDatabase {
   // using InsertExternal().
   virtual void Reset();
 
+  // Replaces the existing set of decoders with the given set. Returns the
+  // payload types that were reassigned or removed while doing so.
+  virtual std::vector<int> SetCodecs(
+      const std::map<int, SdpAudioFormat>& codecs);
+
   // Registers |rtp_payload_type| as a decoder of type |codec_type|. The |name|
   // is only used to populate the name field in the DecoderInfo struct in the
   // database, and can be arbitrary (including empty). Returns kOK on success;
diff --git a/webrtc/modules/audio_coding/neteq/include/neteq.h b/webrtc/modules/audio_coding/neteq/include/neteq.h
index ea05940598..450318e456 100644
--- a/webrtc/modules/audio_coding/neteq/include/neteq.h
+++ b/webrtc/modules/audio_coding/neteq/include/neteq.h
@@ -157,6 +157,9 @@ class NetEq {
   // Returns kOK on success, or kFail in case of an error.
   virtual int GetAudio(AudioFrame* audio_frame, bool* muted) = 0;
 
+  // Replaces the current set of decoders with the given one.
+  virtual void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) = 0;
+
   // Associates |rtp_payload_type| with |codec| and |codec_name|, and stores the
   // information in the codec database. Returns 0 on success, -1 on failure.
   // The name is only used to provide information back to the caller about the
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 5015b7e9fe..501e5672d1 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -212,6 +212,15 @@ int NetEqImpl::GetAudio(AudioFrame* audio_frame, bool* muted) {
   return kOK;
 }
 
+void NetEqImpl::SetCodecs(const std::map<int, SdpAudioFormat>& codecs) {
+  rtc::CritScope lock(&crit_sect_);
+  const std::vector<int> changed_payload_types =
+      decoder_database_->SetCodecs(codecs);
+  for (const int pt : changed_payload_types) {
+    packet_buffer_->DiscardPacketsWithPayloadType(pt);
+  }
+}
+
 int NetEqImpl::RegisterPayloadType(NetEqDecoder codec,
                                    const std::string& name,
                                    uint8_t rtp_payload_type) {
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.h b/webrtc/modules/audio_coding/neteq/neteq_impl.h
index a8c3462708..88c0308f68 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.h
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.h
@@ -111,6 +111,8 @@ class NetEqImpl : public webrtc::NetEq {
 
   int GetAudio(AudioFrame* audio_frame, bool* muted) override;
 
+  void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
+
   int RegisterPayloadType(NetEqDecoder codec,
                           const std::string& codec_name,
                           uint8_t rtp_payload_type) override;
diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn
index 541edd9258..c1d69e07bb 100644
--- a/webrtc/modules/rtp_rtcp/BUILD.gn
+++ b/webrtc/modules/rtp_rtcp/BUILD.gn
@@ -168,6 +168,7 @@ rtc_static_library("rtp_rtcp") {
   deps = [
     "../..:webrtc_common",
     "../../api:transport_api",
+    "../../api/audio_codecs:audio_codecs_api",
     "../../base:gtest_prod",
     "../../base:rtc_base_approved",
     "../../base:rtc_task_queue",
@@ -175,6 +176,7 @@ rtc_static_library("rtp_rtcp") {
     "../../common_video",
     "../../logging:rtc_event_log_api",
     "../../system_wrappers",
+    "../audio_coding:audio_format_conversion",
     "../remote_bitrate_estimator",
   ]
 
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h b/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
index 767bd07f26..029de5dfcc 100644
--- a/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
@@ -15,6 +15,7 @@
 #include <memory>
 #include <set>
 
+#include "webrtc/api/audio_codecs/audio_format.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/deprecation.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
@@ -43,6 +44,10 @@ class RTPPayloadRegistry {
 
   // TODO(magjed): Split RTPPayloadRegistry into separate Audio and Video class
   // and simplify the code. http://crbug/webrtc/6743.
+
+  // Replace all audio receive payload types with the given map.
+  void SetAudioReceivePayloads(std::map<int, SdpAudioFormat> codecs);
+
   int32_t RegisterReceivePayload(const CodecInst& audio_codec,
                                  bool* created_new_payload_type);
   int32_t RegisterReceivePayload(const VideoCodec& video_codec);
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
index 1023bf8bb2..234fd7fce5 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
@@ -16,6 +16,7 @@
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 
 namespace webrtc {
@@ -119,6 +120,31 @@ RTPPayloadRegistry::RTPPayloadRegistry()
 
 RTPPayloadRegistry::~RTPPayloadRegistry() = default;
 
+void RTPPayloadRegistry::SetAudioReceivePayloads(
+    std::map<int, SdpAudioFormat> codecs) {
+  rtc::CritScope cs(&crit_sect_);
+
+#if RTC_DCHECK_IS_ON
+  RTC_DCHECK(!used_for_video_);
+  used_for_audio_ = true;
+#endif
+
+  payload_type_map_.clear();
+  for (const auto& kv : codecs) {
+    const int& rtp_payload_type = kv.first;
+    const SdpAudioFormat& audio_format = kv.second;
+    const CodecInst ci = SdpToCodecInst(rtp_payload_type, audio_format);
+    RTC_DCHECK(IsPayloadTypeValid(rtp_payload_type));
+    payload_type_map_.insert(
+        std::make_pair(rtp_payload_type, CreatePayloadType(ci)));
+  }
+
+  // Clear the value of last received payload type since it might mean
+  // something else now.
+  last_received_payload_type_ = -1;
+  last_received_media_payload_type_ = -1;
+}
+
 int32_t RTPPayloadRegistry::RegisterReceivePayload(const CodecInst& audio_codec,
                                                    bool* created_new_payload) {
   rtc::CritScope cs(&crit_sect_);
diff --git a/webrtc/test/mock_voe_channel_proxy.h b/webrtc/test/mock_voe_channel_proxy.h
index d704e1933b..450dd7bcb7 100644
--- a/webrtc/test/mock_voe_channel_proxy.h
+++ b/webrtc/test/mock_voe_channel_proxy.h
@@ -87,6 +87,8 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
   MOCK_METHOD1(SetSendCodec, bool(const CodecInst& codec_inst));
   MOCK_METHOD2(SetSendCNPayloadType,
                bool(int type, PayloadFrequencies frequency));
+  MOCK_METHOD1(SetReceiveCodecs,
+               void(const std::map<int, SdpAudioFormat>& codecs));
   MOCK_METHOD1(OnTwccBasedUplinkPacketLossRate, void(float packet_loss_rate));
   MOCK_METHOD1(OnRecoverableUplinkPacketLossRate,
                void(float recoverable_packet_loss_rate));
diff --git a/webrtc/test/mock_voice_engine.h b/webrtc/test/mock_voice_engine.h
index ca3fe4f466..6b1f9cf31f 100644
--- a/webrtc/test/mock_voice_engine.h
+++ b/webrtc/test/mock_voice_engine.h
@@ -52,6 +52,11 @@ class MockVoiceEngine : public VoiceEngineImpl {
               new testing::NiceMock<webrtc::test::MockVoEChannelProxy>();
           EXPECT_CALL(*proxy, GetAudioDecoderFactory())
               .WillRepeatedly(testing::ReturnRef(decoder_factory_));
+          EXPECT_CALL(*proxy, SetReceiveCodecs(testing::_))
+              .WillRepeatedly(testing::Invoke(
+                  [](const std::map<int, SdpAudioFormat>& codecs) {
+                    EXPECT_THAT(codecs, testing::IsEmpty());
+                  }));
           return proxy;
         }));
 
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index d2b30ca8f7..3b562f9156 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -987,9 +987,10 @@ int32_t Channel::Init() {
     return -1;
   }
 
-  // --- Register all supported codecs to the receiving side of the
-  // RTP/RTCP module
+  return 0;
+}
 
+void Channel::RegisterLegacyCodecs() {
   CodecInst codec;
   const uint8_t nSupportedCodecs = AudioCodingModule::NumberOfCodecs();
 
@@ -1041,8 +1042,6 @@ int32_t Channel::Init() {
       }
     }
   }
-
-  return 0;
 }
 
 void Channel::Terminate() {
@@ -1360,6 +1359,11 @@ int32_t Channel::GetVADStatus(bool& enabledVAD,
   return 0;
 }
 
+void Channel::SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) {
+  rtp_payload_registry_->SetAudioReceivePayloads(codecs);
+  audio_coding_->SetReceiveCodecs(codecs);
+}
+
 int32_t Channel::SetRecPayloadType(const CodecInst& codec) {
   return SetRecPayloadType(codec.pltype, CodecInstToSdp(codec));
 }
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 0a12f21e92..1365ae843e 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -151,6 +151,7 @@ class Channel
           uint32_t instanceId,
           const VoEBase::ChannelConfig& config);
   int32_t Init();
+  void RegisterLegacyCodecs();
   void Terminate();
   int32_t SetEngineInformation(Statistics& engineStatistics,
                                OutputMixer& outputMixer,
@@ -168,6 +169,8 @@ class Channel
   // go.
   const rtc::scoped_refptr<AudioDecoderFactory>& GetAudioDecoderFactory() const;
 
+  void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
+
   // API methods
 
   // VoEBase
diff --git a/webrtc/voice_engine/channel_proxy.cc b/webrtc/voice_engine/channel_proxy.cc
index 45cbf10819..d67011e680 100644
--- a/webrtc/voice_engine/channel_proxy.cc
+++ b/webrtc/voice_engine/channel_proxy.cc
@@ -173,6 +173,12 @@ void ChannelProxy::SetRecPayloadType(int payload_type,
   RTC_DCHECK_EQ(0, result);
 }
 
+void ChannelProxy::SetReceiveCodecs(
+    const std::map<int, SdpAudioFormat>& codecs) {
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+  channel()->SetReceiveCodecs(codecs);
+}
+
 void ChannelProxy::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   channel()->SetSink(std::move(sink));
@@ -379,6 +385,11 @@ void ChannelProxy::OnRecoverableUplinkPacketLossRate(
   channel()->OnRecoverableUplinkPacketLossRate(recoverable_packet_loss_rate);
 }
 
+void ChannelProxy::RegisterLegacyCodecs() {
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+  channel()->RegisterLegacyCodecs();
+}
+
 Channel* ChannelProxy::channel() const {
   RTC_DCHECK(channel_owner_.channel());
   return channel_owner_.channel();
diff --git a/webrtc/voice_engine/channel_proxy.h b/webrtc/voice_engine/channel_proxy.h
index 685a168845..b808096f45 100644
--- a/webrtc/voice_engine/channel_proxy.h
+++ b/webrtc/voice_engine/channel_proxy.h
@@ -82,6 +82,7 @@ class ChannelProxy {
   virtual void SetBitrate(int bitrate_bps, int64_t probing_interval_ms);
   virtual void SetRecPayloadType(int payload_type,
                                  const SdpAudioFormat& format);
+  virtual void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
   virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink);
   virtual void SetInputMute(bool muted);
   virtual void RegisterExternalTransport(Transport* transport);
@@ -119,6 +120,7 @@ class ChannelProxy {
   virtual void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate);
   virtual void OnRecoverableUplinkPacketLossRate(
       float recoverable_packet_loss_rate);
+  virtual void RegisterLegacyCodecs();
 
  private:
   Channel* channel() const;
diff --git a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
index b15d72fecb..f1eee9c6a2 100644
--- a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
+++ b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc
@@ -15,6 +15,8 @@
 #include "webrtc/base/byteorder.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/voice_engine/channel_proxy.h"
+#include "webrtc/voice_engine/voice_engine_impl.h"
 
 namespace {
   static const unsigned int kReflectorSsrc = 0x0000;
@@ -62,6 +64,9 @@ ConferenceTransport::ConferenceTransport()
 
   EXPECT_EQ(0, local_base_->Init());
   local_sender_ = local_base_->CreateChannel();
+  static_cast<webrtc::VoiceEngineImpl*>(local_voe_)
+      ->GetChannelProxy(local_sender_)
+      ->RegisterLegacyCodecs();
   EXPECT_EQ(0, local_network_->RegisterExternalTransport(local_sender_, *this));
   EXPECT_EQ(0, local_rtp_rtcp_->SetLocalSSRC(local_sender_, kLocalSsrc));
   EXPECT_EQ(0, local_rtp_rtcp_->
@@ -72,6 +77,9 @@ ConferenceTransport::ConferenceTransport()
 
   EXPECT_EQ(0, remote_base_->Init());
   reflector_ = remote_base_->CreateChannel();
+  static_cast<webrtc::VoiceEngineImpl*>(remote_voe_)
+      ->GetChannelProxy(reflector_)
+      ->RegisterLegacyCodecs();
   EXPECT_EQ(0, remote_network_->RegisterExternalTransport(reflector_, *this));
   EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(reflector_, kReflectorSsrc));
 
@@ -222,6 +230,9 @@ void ConferenceTransport::SetRtt(unsigned int rtt_ms) {
 unsigned int ConferenceTransport::AddStream(std::string file_name,
                                             webrtc::FileFormats format) {
   const int new_sender = remote_base_->CreateChannel();
+  static_cast<webrtc::VoiceEngineImpl*>(remote_voe_)
+      ->GetChannelProxy(new_sender)
+      ->RegisterLegacyCodecs();
   EXPECT_EQ(0, remote_network_->RegisterExternalTransport(new_sender, *this));
 
   const unsigned int remote_ssrc = kFirstRemoteSsrc + stream_count_++;
@@ -235,6 +246,9 @@ unsigned int ConferenceTransport::AddStream(std::string file_name,
       new_sender, file_name.c_str(), true, false, format, 1.0));
 
   const int new_receiver = local_base_->CreateChannel();
+  static_cast<webrtc::VoiceEngineImpl*>(local_voe_)
+      ->GetChannelProxy(new_receiver)
+      ->RegisterLegacyCodecs();
   EXPECT_EQ(0, local_base_->AssociateSendChannel(new_receiver, local_sender_));
 
   EXPECT_EQ(0, local_network_->RegisterExternalTransport(new_receiver, *this));
diff --git a/webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.cc b/webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.cc
index 0fbe63550d..5582124a38 100644
--- a/webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.cc
+++ b/webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.cc
@@ -16,5 +16,6 @@ AfterStreamingFixture::AfterStreamingFixture()
   webrtc::VoiceEngineImpl* voe_impl =
       static_cast<webrtc::VoiceEngineImpl*>(voice_engine_);
   channel_proxy_ = voe_impl->GetChannelProxy(channel_);
+  channel_proxy_->RegisterLegacyCodecs();
   ResumePlaying();
 }
diff --git a/webrtc/voice_engine/test/auto_test/standard/codec_before_streaming_test.cc b/webrtc/voice_engine/test/auto_test/standard/codec_before_streaming_test.cc
index a733b12705..d9a5e2b794 100644
--- a/webrtc/voice_engine/test/auto_test/standard/codec_before_streaming_test.cc
+++ b/webrtc/voice_engine/test/auto_test/standard/codec_before_streaming_test.cc
@@ -8,7 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "webrtc/voice_engine/channel_proxy.h"
 #include "webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h"
+#include "webrtc/voice_engine/voice_engine_impl.h"
 
 class CodecBeforeStreamingTest : public AfterInitializationFixture {
  protected:
@@ -19,6 +21,9 @@ class CodecBeforeStreamingTest : public AfterInitializationFixture {
     codec_instance_.pacsize = 480;
 
     channel_ = voe_base_->CreateChannel();
+    static_cast<webrtc::VoiceEngineImpl*>(voice_engine_)
+        ->GetChannelProxy(channel_)
+        ->RegisterLegacyCodecs();
   }
 
   void TearDown() {