diff --git a/.gn b/.gn
index 4aa7a76345..28e8fad9c2 100644
--- a/.gn
+++ b/.gn
@@ -22,6 +22,9 @@ secondary_source = "//build/secondary/"
 check_targets = [
   "//webrtc/api:audio_mixer_api",
   "//webrtc/api:rtc_stats_api",
+  "//webrtc/modules/audio_coding:audio_decoder_factory_interface",
+  "//webrtc/modules/audio_coding:audio_format",
+  "//webrtc/modules/audio_coding:audio_format_conversion",
   "//webrtc/modules/audio_coding:g711_test",
   "//webrtc/modules/audio_coding:g722_test",
   "//webrtc/modules/audio_coding:ilbc_test",
diff --git a/webrtc/modules/BUILD.gn b/webrtc/modules/BUILD.gn
index 6c472a399f..0b08e9079e 100644
--- a/webrtc/modules/BUILD.gn
+++ b/webrtc/modules/BUILD.gn
@@ -63,6 +63,7 @@ if (rtc_include_tests) {
       "..:webrtc_common",
       "../common_video",
       "../modules/audio_coding",
+      "../modules/audio_coding:audio_format_conversion",
       "../modules/rtp_rtcp",
       "../modules/utility",
       "../modules/video_coding",
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 078fb4bccc..6f47dcce94 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -39,14 +39,36 @@ audio_coding_deps = audio_codec_deps + [
                       "../../system_wrappers",
                     ]
 
-rtc_static_library("audio_decoder_factory_interface") {
+rtc_static_library("audio_format") {
   sources = [
-    "codecs/audio_decoder_factory.h",
     "codecs/audio_format.cc",
     "codecs/audio_format.h",
   ]
   deps = [
-    "../..:webrtc_common",
+    "//webrtc:webrtc_common",
+  ]
+}
+
+rtc_static_library("audio_format_conversion") {
+  sources = [
+    "codecs/audio_format_conversion.cc",
+    "codecs/audio_format_conversion.h",
+  ]
+  deps = [
+    ":audio_format",
+    "//webrtc:webrtc_common",
+    "//webrtc/base:rtc_base_approved",
+  ]
+}
+
+rtc_source_set("audio_decoder_factory_interface") {
+  sources = [
+    "codecs/audio_decoder_factory.h",
+  ]
+  deps = [
+    ":audio_decoder_interface",
+    ":audio_format",
+    "//webrtc/base:rtc_base_approved",
   ]
 }
 
@@ -934,6 +956,7 @@ if (rtc_include_tests) {
 
     deps = [
       ":audio_coding",
+      ":audio_format_conversion",
       "../../:webrtc_common",
       "../../system_wrappers",
       "../../system_wrappers:system_wrappers_default",
@@ -959,6 +982,7 @@ if (rtc_include_tests) {
 
     deps = [
       ":audio_coding",
+      ":audio_format_conversion",
       "../../:webrtc_common",
       "../../system_wrappers",
       "../../system_wrappers:system_wrappers_default",
diff --git a/webrtc/modules/audio_coding/acm2/acm_receive_test.cc b/webrtc/modules/audio_coding/acm2/acm_receive_test.cc
index 056ff2273c..c2549323c7 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receive_test.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receive_test.cc
@@ -15,6 +15,7 @@
 
 #include <memory>
 
+#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
@@ -132,7 +133,9 @@ void AcmReceiveTestOldApi::RegisterDefaultCodecs() {
   for (int n = 0; n < acm_->NumberOfCodecs(); n++) {
     ASSERT_EQ(0, acm_->Codec(n, &my_codec_param)) << "Failed to get codec.";
     if (ModifyAndUseThisCodec(&my_codec_param)) {
-      ASSERT_EQ(0, acm_->RegisterReceiveCodec(my_codec_param))
+      ASSERT_EQ(true,
+                acm_->RegisterReceiveCodec(my_codec_param.pltype,
+                                           CodecInstToSdp(my_codec_param)))
           << "Couldn't register receive codec.\n";
     }
   }
@@ -151,22 +154,14 @@ void AcmReceiveTestOldApi::RegisterNetEqTestCodecs() {
                                    my_codec_param.plfreq,
                                    my_codec_param.channels,
                                    &my_codec_param.pltype)) {
-      ASSERT_EQ(0, acm_->RegisterReceiveCodec(my_codec_param))
+      ASSERT_EQ(true,
+                acm_->RegisterReceiveCodec(my_codec_param.pltype,
+                                           CodecInstToSdp(my_codec_param)))
           << "Couldn't register receive codec.\n";
     }
   }
 }
 
-int AcmReceiveTestOldApi::RegisterExternalReceiveCodec(
-    int rtp_payload_type,
-    AudioDecoder* external_decoder,
-    int sample_rate_hz,
-    int num_channels,
-    const std::string& name) {
-  return acm_->RegisterExternalReceiveCodec(rtp_payload_type, external_decoder,
-                                            sample_rate_hz, num_channels, name);
-}
-
 void AcmReceiveTestOldApi::Run() {
   for (std::unique_ptr<Packet> packet(packet_source_->NextPacket()); packet;
        packet = packet_source_->NextPacket()) {
diff --git a/webrtc/modules/audio_coding/acm2/acm_receive_test.h b/webrtc/modules/audio_coding/acm2/acm_receive_test.h
index 3145ad3812..834fa5626f 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receive_test.h
+++ b/webrtc/modules/audio_coding/acm2/acm_receive_test.h
@@ -50,12 +50,6 @@ class AcmReceiveTestOldApi {
   // files.
   void RegisterNetEqTestCodecs();
 
-  int RegisterExternalReceiveCodec(int rtp_payload_type,
-                                   AudioDecoder* external_decoder,
-                                   int sample_rate_hz,
-                                   int num_channels,
-                                   const std::string& name);
-
   // Runs the test and returns true if successful.
   void Run();
 
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index b6fcd01b63..863dcd1177 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -21,6 +21,7 @@
 #include "webrtc/modules/audio_coding/acm2/acm_send_test.h"
 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
+#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
 #include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
 #include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
@@ -777,7 +778,8 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
     // Register iSAC codec in ACM, effectively unregistering the PCM16B codec
     // registered in AudioCodingModuleTestOldApi::SetUp();
     // Only register the decoder for now. The encoder is registered later.
-    ASSERT_EQ(0, acm_->RegisterReceiveCodec(codec_));
+    ASSERT_EQ(true, acm_->RegisterReceiveCodec(codec_.pltype,
+                                               CodecInstToSdp(codec_)));
   }
 
   void StartThreads() {
diff --git a/webrtc/modules/audio_coding/audio_coding.gypi b/webrtc/modules/audio_coding/audio_coding.gypi
index 5f9eb2cdad..a9b9e20e63 100644
--- a/webrtc/modules/audio_coding/audio_coding.gypi
+++ b/webrtc/modules/audio_coding/audio_coding.gypi
@@ -87,6 +87,8 @@
         'codecs/audio_decoder_factory.h',
         'codecs/audio_format.cc',
         'codecs/audio_format.h',
+        'codecs/audio_format_conversion.cc',
+        'codecs/audio_format_conversion.h',
       ],
     },
     {
diff --git a/webrtc/modules/audio_coding/codecs/audio_format_conversion.cc b/webrtc/modules/audio_coding/codecs/audio_format_conversion.cc
new file mode 100644
index 0000000000..ef9aa4479b
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/audio_format_conversion.cc
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/safe_conversions.h"
+
+namespace webrtc {
+
+SdpAudioFormat CodecInstToSdp(const CodecInst& ci) {
+  if (STR_CASE_CMP(ci.plname, "g722") == 0 && ci.plfreq == 16000) {
+    RTC_CHECK(ci.channels == 1 || ci.channels == 2);
+    return {"g722", 8000, static_cast<int>(ci.channels)};
+  } else if (STR_CASE_CMP(ci.plname, "opus") == 0 && ci.plfreq == 48000) {
+    RTC_CHECK(ci.channels == 1 || ci.channels == 2);
+    return {"opus", 48000, 2, {{"stereo", ci.channels == 1 ? "0" : "1"}}};
+  } else {
+    return {ci.plname, ci.plfreq, rtc::checked_cast<int>(ci.channels)};
+  }
+}
+
+}  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/codecs/audio_format_conversion.h b/webrtc/modules/audio_coding/codecs/audio_format_conversion.h
new file mode 100644
index 0000000000..ff71282f7e
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/audio_format_conversion.h
@@ -0,0 +1,23 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_CONVERSION_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_CONVERSION_H_
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/audio_format.h"
+
+namespace webrtc {
+
+SdpAudioFormat CodecInstToSdp(const CodecInst& codec_inst);
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_CONVERSION_H_
diff --git a/webrtc/modules/audio_coding/test/APITest.cc b/webrtc/modules/audio_coding/test/APITest.cc
index ffa675c0fa..326a303ea3 100644
--- a/webrtc/modules/audio_coding/test/APITest.cc
+++ b/webrtc/modules/audio_coding/test/APITest.cc
@@ -23,6 +23,7 @@
 #include "webrtc/base/timeutils.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
@@ -141,7 +142,8 @@ int16_t APITest::SetUp() {
       // Check registration with an already occupied payload type
       int currentPayloadType = dummyCodec.pltype;
       dummyCodec.pltype = 97;  //lastPayloadType;
-      CHECK_ERROR(_acmB->RegisterReceiveCodec(dummyCodec));
+      EXPECT_EQ(true, _acmB->RegisterReceiveCodec(dummyCodec.pltype,
+                                                  CodecInstToSdp(dummyCodec)));
       dummyCodec.pltype = currentPayloadType;
     }
 
@@ -152,7 +154,8 @@ int16_t APITest::SetUp() {
       AudioCodingModule::Codec(n + 1, &nextCodec);
       dummyCodec.pltype = nextCodec.pltype;
       if (!FixedPayloadTypeCodec(nextCodec.plname)) {
-        _acmB->RegisterReceiveCodec(dummyCodec);
+        _acmB->RegisterReceiveCodec(dummyCodec.pltype,
+                                    CodecInstToSdp(dummyCodec));
       }
       dummyCodec.pltype = currentPayloadType;
     }
@@ -163,14 +166,17 @@ int16_t APITest::SetUp() {
       AudioCodingModule::Codec(n + 1, &nextCodec);
       nextCodec.pltype = dummyCodec.pltype;
       if (!FixedPayloadTypeCodec(nextCodec.plname)) {
-        CHECK_ERROR_MT(_acmA->RegisterReceiveCodec(nextCodec));
+        EXPECT_EQ(true, _acmA->RegisterReceiveCodec(nextCodec.pltype,
+                                                    CodecInstToSdp(nextCodec)));
         CHECK_ERROR_MT(_acmA->UnregisterReceiveCodec(nextCodec.pltype));
       }
     }
 
-    CHECK_ERROR_MT(_acmA->RegisterReceiveCodec(dummyCodec));
+    EXPECT_EQ(true, _acmA->RegisterReceiveCodec(dummyCodec.pltype,
+                                                CodecInstToSdp(dummyCodec)));
     printf("   side A done!");
-    CHECK_ERROR_MT(_acmB->RegisterReceiveCodec(dummyCodec));
+    EXPECT_EQ(true, _acmB->RegisterReceiveCodec(dummyCodec.pltype,
+                                                CodecInstToSdp(dummyCodec)));
     printf("   side B done!\n");
 
     if (!strcmp(dummyCodec.plname, "CN")) {
@@ -871,7 +877,8 @@ void APITest::TestRegisteration(char sendSide) {
                   "Register receive codec with default Payload, AUDIO BACK.\n");
           fflush (stdout);
         }
-        CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(*myCodec));
+        EXPECT_EQ(true, receiveACM->RegisterReceiveCodec(
+                            myCodec->pltype, CodecInstToSdp(*myCodec)));
         //CHECK_ERROR_MT(sendACM->RegisterSendCodec(*myCodec));
         myEvent->Wait(20);
         {
@@ -884,7 +891,8 @@ void APITest::TestRegisteration(char sendSide) {
       }
     }
     if (i == 32) {
-      CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(*myCodec));
+      EXPECT_EQ(true, receiveACM->RegisterReceiveCodec(
+                          myCodec->pltype, CodecInstToSdp(*myCodec)));
       {
         WriteLockScoped wl(_apiTestRWLock);
         *thereIsDecoder = true;
@@ -896,7 +904,8 @@ void APITest::TestRegisteration(char sendSide) {
               "Register receive codec with fixed Payload, AUDIO BACK.\n");
       fflush (stdout);
     }
-    CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(*myCodec));
+    EXPECT_EQ(true, receiveACM->RegisterReceiveCodec(myCodec->pltype,
+                                                     CodecInstToSdp(*myCodec)));
     //CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec->pltype));
     //CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(*myCodec));
     myEvent->Wait(20);
diff --git a/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc b/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
index 2bcae5a425..3eada25c8c 100644
--- a/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -17,6 +17,7 @@
 
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/system_wrappers/include/trace.h"
@@ -132,11 +133,13 @@ void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
   for (int i = 0; i < noOfCodecs; i++) {
     EXPECT_EQ(0, acm->Codec(i, &recvCodec));
     if (recvCodec.channels == channels)
-      EXPECT_EQ(0, acm->RegisterReceiveCodec(recvCodec));
+      EXPECT_EQ(true, acm->RegisterReceiveCodec(recvCodec.pltype,
+                                                CodecInstToSdp(recvCodec)));
     // Forces mono/stereo for Opus.
     if (!strcmp(recvCodec.plname, "opus")) {
       recvCodec.channels = channels;
-      EXPECT_EQ(0, acm->RegisterReceiveCodec(recvCodec));
+      EXPECT_EQ(true, acm->RegisterReceiveCodec(recvCodec.pltype,
+                                                CodecInstToSdp(recvCodec)));
     }
   }
 
diff --git a/webrtc/modules/audio_coding/test/TestAllCodecs.cc b/webrtc/modules/audio_coding/test/TestAllCodecs.cc
index e5d9784e9c..6753d8e2c7 100644
--- a/webrtc/modules/audio_coding/test/TestAllCodecs.cc
+++ b/webrtc/modules/audio_coding/test/TestAllCodecs.cc
@@ -15,6 +15,7 @@
 #include <string>
 
 #include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/test/utility.h"
@@ -141,7 +142,8 @@ void TestAllCodecs::Perform() {
     if (!strcmp(my_codec_param.plname, "opus")) {
       my_codec_param.channels = 1;
     }
-    acm_b_->RegisterReceiveCodec(my_codec_param);
+    acm_b_->RegisterReceiveCodec(my_codec_param.pltype,
+                                 CodecInstToSdp(my_codec_param));
   }
 
   // Create and connect the channel
diff --git a/webrtc/modules/audio_coding/test/TestRedFec.cc b/webrtc/modules/audio_coding/test/TestRedFec.cc
index 4e8b123b78..7c252a13d9 100644
--- a/webrtc/modules/audio_coding/test/TestRedFec.cc
+++ b/webrtc/modules/audio_coding/test/TestRedFec.cc
@@ -13,6 +13,7 @@
 #include <assert.h>
 
 #include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/system_wrappers/include/trace.h"
@@ -77,7 +78,8 @@ void TestRedFec::Perform() {
     if (!strcmp(myCodecParam.plname, "opus")) {
       myCodecParam.channels = 1;
     }
-    EXPECT_EQ(0, _acmB->RegisterReceiveCodec(myCodecParam));
+    EXPECT_EQ(true, _acmB->RegisterReceiveCodec(myCodecParam.pltype,
+                                                CodecInstToSdp(myCodecParam)));
   }
 
   // Create and connect the channel
diff --git a/webrtc/modules/audio_coding/test/TestStereo.cc b/webrtc/modules/audio_coding/test/TestStereo.cc
index 572a08c21e..57b7a54c87 100644
--- a/webrtc/modules/audio_coding/test/TestStereo.cc
+++ b/webrtc/modules/audio_coding/test/TestStereo.cc
@@ -15,6 +15,7 @@
 #include <string>
 
 #include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/system_wrappers/include/trace.h"
@@ -171,7 +172,8 @@ void TestStereo::Perform() {
   CodecInst my_codec_param;
   for (uint8_t n = 0; n < num_encoders; n++) {
     EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param));
-    EXPECT_EQ(0, acm_b_->RegisterReceiveCodec(my_codec_param));
+    EXPECT_EQ(true, acm_b_->RegisterReceiveCodec(
+                        my_codec_param.pltype, CodecInstToSdp(my_codec_param)));
   }
 
   // Test that unregister all receive codecs works.
@@ -183,7 +185,8 @@ void TestStereo::Perform() {
   // Register all available codes as receiving codecs once more.
   for (uint8_t n = 0; n < num_encoders; n++) {
     EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param));
-    EXPECT_EQ(0, acm_b_->RegisterReceiveCodec(my_codec_param));
+    EXPECT_EQ(true, acm_b_->RegisterReceiveCodec(
+                        my_codec_param.pltype, CodecInstToSdp(my_codec_param)));
   }
 
   // Create and connect the channel.
@@ -597,7 +600,9 @@ void TestStereo::Perform() {
     EXPECT_EQ(0, acm_b_->Codec(n, &opus_codec_param));
     if (!strcmp(opus_codec_param.plname, "opus")) {
       opus_codec_param.channels = 1;
-      EXPECT_EQ(0, acm_b_->RegisterReceiveCodec(opus_codec_param));
+      EXPECT_EQ(true,
+                acm_b_->RegisterReceiveCodec(opus_codec_param.pltype,
+                                             CodecInstToSdp(opus_codec_param)));
       break;
     }
   }
@@ -630,7 +635,9 @@ void TestStereo::Perform() {
         " Decode: stereo\n", test_cntr_);
   }
   opus_codec_param.channels = 2;
-  EXPECT_EQ(0, acm_b_->RegisterReceiveCodec(opus_codec_param));
+  EXPECT_EQ(true,
+            acm_b_->RegisterReceiveCodec(opus_codec_param.pltype,
+                                         CodecInstToSdp(opus_codec_param)));
   Run(channel_a2b_, audio_channels, 2);
   out_file_.Close();
   // Decode in mono.
@@ -642,7 +649,9 @@ void TestStereo::Perform() {
         " Decode: mono\n", test_cntr_);
   }
   opus_codec_param.channels = 1;
-  EXPECT_EQ(0, acm_b_->RegisterReceiveCodec(opus_codec_param));
+  EXPECT_EQ(true,
+            acm_b_->RegisterReceiveCodec(opus_codec_param.pltype,
+                                         CodecInstToSdp(opus_codec_param)));
   Run(channel_a2b_, audio_channels, codec_channels);
   out_file_.Close();
 
diff --git a/webrtc/modules/audio_coding/test/TestVADDTX.cc b/webrtc/modules/audio_coding/test/TestVADDTX.cc
index 505eb13285..dd5b3a76e9 100644
--- a/webrtc/modules/audio_coding/test/TestVADDTX.cc
+++ b/webrtc/modules/audio_coding/test/TestVADDTX.cc
@@ -12,6 +12,7 @@
 
 #include <string>
 
+#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_coding/test/PCMFile.h"
 #include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/test/testsupport/fileutils.h"
@@ -73,7 +74,8 @@ TestVadDtx::TestVadDtx()
 void TestVadDtx::RegisterCodec(CodecInst codec_param) {
   // Set the codec for sending and receiving.
   EXPECT_EQ(0, acm_send_->RegisterSendCodec(codec_param));
-  EXPECT_EQ(0, acm_receive_->RegisterReceiveCodec(codec_param));
+  EXPECT_EQ(true, acm_receive_->RegisterReceiveCodec(
+                      codec_param.pltype, CodecInstToSdp(codec_param)));
   channel_->SetIsStereo(codec_param.channels > 1);
 }
 
diff --git a/webrtc/modules/audio_coding/test/TwoWayCommunication.cc b/webrtc/modules/audio_coding/test/TwoWayCommunication.cc
index ca2a2124ae..7f2af7d716 100644
--- a/webrtc/modules/audio_coding/test/TwoWayCommunication.cc
+++ b/webrtc/modules/audio_coding/test/TwoWayCommunication.cc
@@ -21,6 +21,7 @@
 #endif
 
 #include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/audio_coding/test/PCMFile.h"
 #include "webrtc/modules/audio_coding/test/utility.h"
@@ -97,18 +98,22 @@ void TwoWayCommunication::SetUp() {
 
   //--- Set A codecs
   EXPECT_EQ(0, _acmA->RegisterSendCodec(codecInst_A));
-  EXPECT_EQ(0, _acmA->RegisterReceiveCodec(codecInst_B));
+  EXPECT_EQ(true, _acmA->RegisterReceiveCodec(codecInst_B.pltype,
+                                              CodecInstToSdp(codecInst_B)));
   //--- Set ref-A codecs
   EXPECT_EQ(0, _acmRefA->RegisterSendCodec(codecInst_A));
-  EXPECT_EQ(0, _acmRefA->RegisterReceiveCodec(codecInst_B));
+  EXPECT_EQ(true, _acmRefA->RegisterReceiveCodec(codecInst_B.pltype,
+                                                 CodecInstToSdp(codecInst_B)));
 
   //--- Set B codecs
   EXPECT_EQ(0, _acmB->RegisterSendCodec(codecInst_B));
-  EXPECT_EQ(0, _acmB->RegisterReceiveCodec(codecInst_A));
+  EXPECT_EQ(true, _acmB->RegisterReceiveCodec(codecInst_A.pltype,
+                                              CodecInstToSdp(codecInst_A)));
 
   //--- Set ref-B codecs
   EXPECT_EQ(0, _acmRefB->RegisterSendCodec(codecInst_B));
-  EXPECT_EQ(0, _acmRefB->RegisterReceiveCodec(codecInst_A));
+  EXPECT_EQ(true, _acmRefB->RegisterReceiveCodec(codecInst_A.pltype,
+                                                 CodecInstToSdp(codecInst_A)));
 
   uint16_t frequencyHz;
 
@@ -174,19 +179,23 @@ void TwoWayCommunication::SetUpAutotest() {
 
   //--- Set A codecs
   EXPECT_EQ(0, _acmA->RegisterSendCodec(codecInst_A));
-  EXPECT_EQ(0, _acmA->RegisterReceiveCodec(codecInst_B));
+  EXPECT_EQ(true, _acmA->RegisterReceiveCodec(codecInst_B.pltype,
+                                              CodecInstToSdp(codecInst_B)));
 
   //--- Set ref-A codecs
   EXPECT_GT(_acmRefA->RegisterSendCodec(codecInst_A), -1);
-  EXPECT_GT(_acmRefA->RegisterReceiveCodec(codecInst_B), -1);
+  EXPECT_EQ(true, _acmRefA->RegisterReceiveCodec(codecInst_B.pltype,
+                                                 CodecInstToSdp(codecInst_B)));
 
   //--- Set B codecs
   EXPECT_GT(_acmB->RegisterSendCodec(codecInst_B), -1);
-  EXPECT_GT(_acmB->RegisterReceiveCodec(codecInst_A), -1);
+  EXPECT_EQ(true, _acmB->RegisterReceiveCodec(codecInst_A.pltype,
+                                              CodecInstToSdp(codecInst_A)));
 
   //--- Set ref-B codecs
   EXPECT_EQ(0, _acmRefB->RegisterSendCodec(codecInst_B));
-  EXPECT_EQ(0, _acmRefB->RegisterReceiveCodec(codecInst_A));
+  EXPECT_EQ(true, _acmRefB->RegisterReceiveCodec(codecInst_A.pltype,
+                                                 CodecInstToSdp(codecInst_A)));
 
   uint16_t frequencyHz;
 
@@ -292,7 +301,8 @@ void TwoWayCommunication::Perform() {
       EXPECT_EQ(0, _acmA->InitializeReceiver());
     // Re-register codec on side A.
     if (((secPassed % 7) == 6) && (msecPassed >= 990)) {
-      EXPECT_EQ(0, _acmA->RegisterReceiveCodec(*codecInst_B));
+      EXPECT_EQ(true, _acmA->RegisterReceiveCodec(
+                          codecInst_B->pltype, CodecInstToSdp(*codecInst_B)));
     }
   }
 }
diff --git a/webrtc/modules/audio_coding/test/delay_test.cc b/webrtc/modules/audio_coding/test/delay_test.cc
index 99ec0b82ea..9adeca1e5d 100644
--- a/webrtc/modules/audio_coding/test/delay_test.cc
+++ b/webrtc/modules/audio_coding/test/delay_test.cc
@@ -17,6 +17,7 @@
 #include "gflags/gflags.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/test/Channel.h"
@@ -107,8 +108,9 @@ class DelayTest {
         continue;
       if (STR_CASE_CMP(my_codec_param.plname, "telephone-event") == 0)
         continue;
-      ASSERT_EQ(0, acm_b_->RegisterReceiveCodec(my_codec_param)) <<
-          "Couldn't register receive codec.\n";
+      ASSERT_EQ(true,
+                acm_b_->RegisterReceiveCodec(my_codec_param.pltype,
+                                             CodecInstToSdp(my_codec_param)));
     }
 
     // Create and connect the channel
diff --git a/webrtc/modules/audio_coding/test/iSACTest.cc b/webrtc/modules/audio_coding/test/iSACTest.cc
index 9729cf817a..6ccdf3c3cf 100644
--- a/webrtc/modules/audio_coding/test/iSACTest.cc
+++ b/webrtc/modules/audio_coding/test/iSACTest.cc
@@ -24,6 +24,7 @@
 #endif
 
 #include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_coding/test/utility.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
@@ -94,10 +95,14 @@ void ISACTest::Setup() {
   }
 
   // Register both iSAC-wb & iSAC-swb in both sides as receiver codecs.
-  EXPECT_EQ(0, _acmA->RegisterReceiveCodec(_paramISAC16kHz));
-  EXPECT_EQ(0, _acmA->RegisterReceiveCodec(_paramISAC32kHz));
-  EXPECT_EQ(0, _acmB->RegisterReceiveCodec(_paramISAC16kHz));
-  EXPECT_EQ(0, _acmB->RegisterReceiveCodec(_paramISAC32kHz));
+  EXPECT_EQ(true, _acmA->RegisterReceiveCodec(_paramISAC16kHz.pltype,
+                                              CodecInstToSdp(_paramISAC16kHz)));
+  EXPECT_EQ(true, _acmA->RegisterReceiveCodec(_paramISAC32kHz.pltype,
+                                              CodecInstToSdp(_paramISAC32kHz)));
+  EXPECT_EQ(true, _acmB->RegisterReceiveCodec(_paramISAC16kHz.pltype,
+                                              CodecInstToSdp(_paramISAC16kHz)));
+  EXPECT_EQ(true, _acmB->RegisterReceiveCodec(_paramISAC32kHz.pltype,
+                                              CodecInstToSdp(_paramISAC32kHz)));
 
   //--- Set A-to-B channel
   _channel_A2B.reset(new Channel);
diff --git a/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc b/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc
index ca0a240aec..44ef9df7d9 100644
--- a/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc
+++ b/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc
@@ -14,6 +14,7 @@
 
 #include "gflags/gflags.h"
 #include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/test/Channel.h"
 #include "webrtc/modules/audio_coding/test/PCMFile.h"
@@ -94,7 +95,8 @@ class InsertPacketWithTiming {
                              FLAGS_codec_channels));
     ASSERT_EQ(0, receive_acm_->InitializeReceiver());
     ASSERT_EQ(0, send_acm_->RegisterSendCodec(codec));
-    ASSERT_EQ(0, receive_acm_->RegisterReceiveCodec(codec));
+    ASSERT_EQ(true, receive_acm_->RegisterReceiveCodec(codec.pltype,
+                                                       CodecInstToSdp(codec)));
 
     // Set codec-dependent parameters.
     samples_in_1ms_ = codec.plfreq / 1000;
diff --git a/webrtc/modules/audio_coding/test/opus_test.cc b/webrtc/modules/audio_coding/test/opus_test.cc
index 4b3cb6ba5a..bb595615d8 100644
--- a/webrtc/modules/audio_coding/test/opus_test.cc
+++ b/webrtc/modules/audio_coding/test/opus_test.cc
@@ -15,6 +15,7 @@
 #include <string>
 
 #include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/test/TestStereo.h"
@@ -94,7 +95,9 @@ void OpusTest::Perform() {
   int codec_id = acm_receiver_->Codec("opus", 48000, 2);
   EXPECT_EQ(0, acm_receiver_->Codec(codec_id, &opus_codec_param));
   payload_type_ = opus_codec_param.pltype;
-  EXPECT_EQ(0, acm_receiver_->RegisterReceiveCodec(opus_codec_param));
+  EXPECT_EQ(true,
+            acm_receiver_->RegisterReceiveCodec(
+                opus_codec_param.pltype, CodecInstToSdp(opus_codec_param)));
 
   // Create and connect the channel.
   channel_a2b_ = new TestPackStereo;
@@ -159,7 +162,9 @@ void OpusTest::Perform() {
 
   // Register Opus mono as receiving codec.
   opus_codec_param.channels = 1;
-  EXPECT_EQ(0, acm_receiver_->RegisterReceiveCodec(opus_codec_param));
+  EXPECT_EQ(true,
+            acm_receiver_->RegisterReceiveCodec(
+                opus_codec_param.pltype, CodecInstToSdp(opus_codec_param)));
 
   // Run Opus with 2.5 ms frame size.
   Run(channel_a2b_, audio_channels, 32000, 120);
diff --git a/webrtc/modules/audio_coding/test/target_delay_unittest.cc b/webrtc/modules/audio_coding/test/target_delay_unittest.cc
index 2939453e4f..1b400262ae 100644
--- a/webrtc/modules/audio_coding/test/target_delay_unittest.cc
+++ b/webrtc/modules/audio_coding/test/target_delay_unittest.cc
@@ -30,12 +30,12 @@ class TargetDelayTest : public ::testing::Test {
   void SetUp() {
     EXPECT_TRUE(acm_.get() != NULL);
 
-    CodecInst codec;
-    ASSERT_EQ(0, AudioCodingModule::Codec("L16", &codec, kSampleRateHz, 1));
     ASSERT_EQ(0, acm_->InitializeReceiver());
-    ASSERT_EQ(0, acm_->RegisterReceiveCodec(codec));
+    constexpr int pltype = 108;
+    ASSERT_EQ(true,
+              acm_->RegisterReceiveCodec(pltype, {"L16", kSampleRateHz, 1}));
 
-    rtp_info_.header.payloadType = codec.pltype;
+    rtp_info_.header.payloadType = pltype;
     rtp_info_.header.timestamp = 0;
     rtp_info_.header.ssrc = 0x12345678;
     rtp_info_.header.markerBit = false;
diff --git a/webrtc/modules/utility/source/coder.cc b/webrtc/modules/utility/source/coder.cc
index f72d03b887..71f969097f 100644
--- a/webrtc/modules/utility/source/coder.cc
+++ b/webrtc/modules/utility/source/coder.cc
@@ -9,6 +9,7 @@
  */
 
 #include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/utility/source/coder.h"
@@ -45,9 +46,8 @@ int32_t AudioCoder::SetEncodeCodec(const CodecInst& codec_inst) {
 }
 
 int32_t AudioCoder::SetDecodeCodec(const CodecInst& codec_inst) {
-  if (acm_->RegisterReceiveCodec(codec_inst, [&] {
-        return rent_a_codec_.RentIsacDecoder(codec_inst.plfreq);
-      }) == -1) {
+  if (!acm_->RegisterReceiveCodec(codec_inst.pltype,
+                                  CodecInstToSdp(codec_inst))) {
     return -1;
   }
   memcpy(&receive_codec_, &codec_inst, sizeof(CodecInst));
diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn
index 2b5d029b2f..9a3894859c 100644
--- a/webrtc/voice_engine/BUILD.gn
+++ b/webrtc/voice_engine/BUILD.gn
@@ -93,6 +93,7 @@ rtc_static_library("voice_engine") {
     "../common_audio",
     "../logging:rtc_event_log_api",
     "../modules/audio_coding:audio_decoder_factory_interface",
+    "../modules/audio_coding:audio_format_conversion",
     "../modules/audio_coding:builtin_audio_decoder_factory",
     "../modules/audio_coding:rent_a_codec",
     "../modules/audio_conference_mixer",
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 6d053323da..deea057727 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -22,6 +22,7 @@
 #include "webrtc/base/timeutils.h"
 #include "webrtc/config.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
+#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/include/module_common_types.h"
@@ -48,14 +49,6 @@ namespace {
 constexpr int64_t kMaxRetransmissionWindowMs = 1000;
 constexpr int64_t kMinRetransmissionWindowMs = 30;
 
-bool RegisterReceiveCodec(std::unique_ptr<AudioCodingModule>* acm,
-                          acm2::RentACodec* rac,
-                          const CodecInst& ci) {
-  const int result = (*acm)->RegisterReceiveCodec(
-      ci, [&] { return rac->RentIsacDecoder(ci.plfreq); });
-  return result == 0;
-}
-
 }  // namespace
 
 const int kTelephoneEventAttenuationdB = 10;
@@ -498,7 +491,8 @@ int32_t Channel::OnInitializeDecoder(
   receiveCodec.pacsize = dummyCodec.pacsize;
 
   // Register the new codec to the ACM
-  if (!RegisterReceiveCodec(&audio_coding_, &rent_a_codec_, receiveCodec)) {
+  if (!audio_coding_->RegisterReceiveCodec(receiveCodec.pltype,
+                                           CodecInstToSdp(receiveCodec))) {
     WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
                  "Channel::OnInitializeDecoder() invalid codec ("
                  "pt=%d, name=%s) received - 1",
@@ -1066,7 +1060,8 @@ int32_t Channel::Init() {
     // Register default PT for outband 'telephone-event'
     if (!STR_CASE_CMP(codec.plname, "telephone-event")) {
       if (_rtpRtcpModule->RegisterSendPayload(codec) == -1 ||
-          !RegisterReceiveCodec(&audio_coding_, &rent_a_codec_, codec)) {
+          !audio_coding_->RegisterReceiveCodec(codec.pltype,
+                                               CodecInstToSdp(codec))) {
         WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
                      "Channel::Init() failed to register outband "
                      "'telephone-event' (%d/%d) correctly",
@@ -1077,7 +1072,8 @@ int32_t Channel::Init() {
     if (!STR_CASE_CMP(codec.plname, "CN")) {
       if (!codec_manager_.RegisterEncoder(codec) ||
           !codec_manager_.MakeEncoder(&rent_a_codec_, audio_coding_.get()) ||
-          !RegisterReceiveCodec(&audio_coding_, &rent_a_codec_, codec) ||
+          !audio_coding_->RegisterReceiveCodec(codec.pltype,
+                                               CodecInstToSdp(codec)) ||
           _rtpRtcpModule->RegisterSendPayload(codec) == -1) {
         WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
                      "Channel::Init() failed to register CN (%d/%d) "
@@ -1425,9 +1421,11 @@ int32_t Channel::SetRecPayloadType(const CodecInst& codec) {
       return -1;
     }
   }
-  if (!RegisterReceiveCodec(&audio_coding_, &rent_a_codec_, codec)) {
+  if (!audio_coding_->RegisterReceiveCodec(codec.pltype,
+                                           CodecInstToSdp(codec))) {
     audio_coding_->UnregisterReceiveCodec(codec.pltype);
-    if (!RegisterReceiveCodec(&audio_coding_, &rent_a_codec_, codec)) {
+    if (!audio_coding_->RegisterReceiveCodec(codec.pltype,
+                                             CodecInstToSdp(codec))) {
       _engineStatisticsPtr->SetLastError(
           VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
           "SetRecPayloadType() ACM registration failed - 1");