diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
index eca909cc4..9b61d3348 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
@@ -94,4 +94,8 @@ bool AudioCodingModule::IsCodecValid(const CodecInst& codec) {
   }
 }
 
+AudioCoding* AudioCoding::Create(const Config& config) {
+  return new AudioCodingImpl(config);
+}
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index 381f02025..98e2558d3 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -14,6 +14,7 @@
 #include <stdlib.h>
 #include <vector>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
@@ -2053,4 +2054,255 @@ void AudioCodingModuleImpl::GetDecodingCallStatistics(
 
 }  // namespace acm2
 
+bool AudioCodingImpl::RegisterSendCodec(AudioEncoder* send_codec) {
+  FATAL() << "Not implemented yet.";
+}
+
+bool AudioCodingImpl::RegisterSendCodec(int encoder_type,
+                                        uint8_t payload_type,
+                                        int frame_size_samples) {
+  std::string codec_name;
+  int sample_rate_hz;
+  int channels;
+  if (!MapCodecTypeToParameters(
+          encoder_type, &codec_name, &sample_rate_hz, &channels)) {
+    return false;
+  }
+  webrtc::CodecInst codec;
+  AudioCodingModule::Codec(
+      codec_name.c_str(), &codec, sample_rate_hz, channels);
+  codec.pltype = payload_type;
+  if (frame_size_samples > 0) {
+    codec.pacsize = frame_size_samples;
+  }
+  return acm_old_->RegisterSendCodec(codec) == 0;
+}
+
+const AudioEncoder* AudioCodingImpl::GetSenderInfo() const {
+  FATAL() << "Not implemented yet.";
+}
+
+int AudioCodingImpl::Add10MsAudio(const AudioFrame& audio_frame) {
+  if (acm_old_->Add10MsData(audio_frame) != 0) {
+    return -1;
+  }
+  return acm_old_->Process();
+}
+
+const ReceiverInfo* AudioCodingImpl::GetReceiverInfo() const {
+  FATAL() << "Not implemented yet.";
+}
+
+bool AudioCodingImpl::RegisterReceiveCodec(AudioDecoder* receive_codec) {
+  FATAL() << "Not implemented yet.";
+}
+
+bool AudioCodingImpl::RegisterReceiveCodec(int decoder_type,
+                                           uint8_t payload_type) {
+  std::string codec_name;
+  int sample_rate_hz;
+  int channels;
+  if (!MapCodecTypeToParameters(
+          decoder_type, &codec_name, &sample_rate_hz, &channels)) {
+    return false;
+  }
+  webrtc::CodecInst codec;
+  AudioCodingModule::Codec(
+      codec_name.c_str(), &codec, sample_rate_hz, channels);
+  codec.pltype = payload_type;
+  return acm_old_->RegisterReceiveCodec(codec) == 0;
+}
+
+bool AudioCodingImpl::InsertPacket(const uint8_t* incoming_payload,
+                                   int32_t payload_len_bytes,
+                                   const WebRtcRTPHeader& rtp_info) {
+  return acm_old_->IncomingPacket(
+             incoming_payload, payload_len_bytes, rtp_info) == 0;
+}
+
+bool AudioCodingImpl::InsertPayload(const uint8_t* incoming_payload,
+                                    int32_t payload_len_byte,
+                                    uint8_t payload_type,
+                                    uint32_t timestamp) {
+  FATAL() << "Not implemented yet.";
+}
+
+bool AudioCodingImpl::SetMinimumPlayoutDelay(int time_ms) {
+  FATAL() << "Not implemented yet.";
+}
+
+bool AudioCodingImpl::SetMaximumPlayoutDelay(int time_ms) {
+  FATAL() << "Not implemented yet.";
+}
+
+int AudioCodingImpl::LeastRequiredDelayMs() const {
+  FATAL() << "Not implemented yet.";
+}
+
+bool AudioCodingImpl::PlayoutTimestamp(uint32_t* timestamp) {
+  FATAL() << "Not implemented yet.";
+}
+
+bool AudioCodingImpl::Get10MsAudio(AudioFrame* audio_frame) {
+  return acm_old_->PlayoutData10Ms(playout_frequency_hz_, audio_frame) == 0;
+}
+
+bool AudioCodingImpl::NetworkStatistics(
+    ACMNetworkStatistics* network_statistics) {
+  FATAL() << "Not implemented yet.";
+}
+
+bool AudioCodingImpl::EnableNack(size_t max_nack_list_size) {
+  FATAL() << "Not implemented yet.";
+}
+
+void AudioCodingImpl::DisableNack() {
+  FATAL() << "Not implemented yet.";
+}
+
+std::vector<uint16_t> AudioCodingImpl::GetNackList(
+    int round_trip_time_ms) const {
+  return acm_old_->GetNackList(round_trip_time_ms);
+}
+
+void AudioCodingImpl::GetDecodingCallStatistics(
+    AudioDecodingCallStats* call_stats) const {
+  acm_old_->GetDecodingCallStatistics(call_stats);
+}
+
+bool AudioCodingImpl::MapCodecTypeToParameters(int codec_type,
+                                               std::string* codec_name,
+                                               int* sample_rate_hz,
+                                               int* channels) {
+  switch (codec_type) {
+#ifdef WEBRTC_CODEC_PCM16
+    case acm2::ACMCodecDB::kPCM16B:
+      *codec_name = "L16";
+      *sample_rate_hz = 8000;
+      *channels = 1;
+      break;
+    case acm2::ACMCodecDB::kPCM16Bwb:
+      *codec_name = "L16";
+      *sample_rate_hz = 16000;
+      *channels = 1;
+      break;
+    case acm2::ACMCodecDB::kPCM16Bswb32kHz:
+      *codec_name = "L16";
+      *sample_rate_hz = 32000;
+      *channels = 1;
+      break;
+    case acm2::ACMCodecDB::kPCM16B_2ch:
+      *codec_name = "L16";
+      *sample_rate_hz = 8000;
+      *channels = 2;
+      break;
+    case acm2::ACMCodecDB::kPCM16Bwb_2ch:
+      *codec_name = "L16";
+      *sample_rate_hz = 16000;
+      *channels = 2;
+      break;
+    case acm2::ACMCodecDB::kPCM16Bswb32kHz_2ch:
+      *codec_name = "L16";
+      *sample_rate_hz = 32000;
+      *channels = 2;
+      break;
+#endif
+#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
+    case acm2::ACMCodecDB::kISAC:
+      *codec_name = "ISAC";
+      *sample_rate_hz = 16000;
+      *channels = 1;
+      break;
+#endif
+#ifdef WEBRTC_CODEC_ISAC
+    case acm2::ACMCodecDB::kISACSWB:
+      *codec_name = "ISAC";
+      *sample_rate_hz = 32000;
+      *channels = 1;
+      break;
+    case acm2::ACMCodecDB::kISACFB:
+      *codec_name = "ISAC";
+      *sample_rate_hz = 48000;
+      *channels = 1;
+      break;
+#endif
+#ifdef WEBRTC_CODEC_ILBC
+    case acm2::ACMCodecDB::kILBC:
+      *codec_name = "ILBC";
+      *sample_rate_hz = 8000;
+      *channels = 1;
+      break;
+#endif
+    case acm2::ACMCodecDB::kPCMA:
+      *codec_name = "PCMA";
+      *sample_rate_hz = 8000;
+      *channels = 1;
+      break;
+    case acm2::ACMCodecDB::kPCMA_2ch:
+      *codec_name = "PCMA";
+      *sample_rate_hz = 8000;
+      *channels = 2;
+      break;
+    case acm2::ACMCodecDB::kPCMU:
+      *codec_name = "PCMU";
+      *sample_rate_hz = 8000;
+      *channels = 1;
+      break;
+    case acm2::ACMCodecDB::kPCMU_2ch:
+      *codec_name = "PCMU";
+      *sample_rate_hz = 8000;
+      *channels = 2;
+      break;
+#ifdef WEBRTC_CODEC_G722
+    case acm2::ACMCodecDB::kG722:
+      *codec_name = "G722";
+      *sample_rate_hz = 16000;
+      *channels = 1;
+      break;
+    case acm2::ACMCodecDB::kG722_2ch:
+      *codec_name = "G722";
+      *sample_rate_hz = 16000;
+      *channels = 2;
+      break;
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+    case acm2::ACMCodecDB::kOpus:
+      *codec_name = "opus";
+      *sample_rate_hz = 48000;
+      *channels = 2;
+      break;
+#endif
+    case acm2::ACMCodecDB::kCNNB:
+      *codec_name = "CN";
+      *sample_rate_hz = 8000;
+      *channels = 1;
+      break;
+    case acm2::ACMCodecDB::kCNWB:
+      *codec_name = "CN";
+      *sample_rate_hz = 16000;
+      *channels = 1;
+      break;
+    case acm2::ACMCodecDB::kCNSWB:
+      *codec_name = "CN";
+      *sample_rate_hz = 32000;
+      *channels = 1;
+      break;
+    case acm2::ACMCodecDB::kRED:
+      *codec_name = "red";
+      *sample_rate_hz = 8000;
+      *channels = 1;
+      break;
+#ifdef WEBRTC_CODEC_AVT
+    case acm2::ACMCodecDB::kAVT:
+      *codec_name = "telephone-event";
+      *sample_rate_hz = 8000;
+      *channels = 1;
+      break;
+#endif
+    default:
+      FATAL() << "Codec type " << codec_type << " not supported.";
+  }
+  return true;
+}
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
index 635b4dcbd..41c28949e 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
@@ -382,6 +382,88 @@ class AudioCodingModuleImpl : public AudioCodingModule {
 
 }  // namespace acm2
 
+class AudioCodingImpl : public AudioCoding {
+ public:
+  AudioCodingImpl(const Config& config) {
+    AudioCodingModule::Config config_old;
+    config_old.id = 0;
+    config_old.neteq_config = config.neteq_config;
+    config_old.clock = config.clock;
+    acm_old_.reset(new acm2::AudioCodingModuleImpl(config_old));
+    acm_old_->RegisterTransportCallback(config.transport);
+    acm_old_->RegisterVADCallback(config.vad_callback);
+    acm_old_->SetDtmfPlayoutStatus(config.play_dtmf);
+    if (config.initial_playout_delay_ms > 0) {
+      acm_old_->SetInitialPlayoutDelay(config.initial_playout_delay_ms);
+    }
+    playout_frequency_hz_ = config.playout_frequency_hz;
+  }
+
+  virtual ~AudioCodingImpl() OVERRIDE {};
+
+  virtual bool RegisterSendCodec(AudioEncoder* send_codec) OVERRIDE;
+
+  virtual bool RegisterSendCodec(int encoder_type,
+                                 uint8_t payload_type,
+                                 int frame_size_samples = 0) OVERRIDE;
+
+  virtual const AudioEncoder* GetSenderInfo() const OVERRIDE;
+
+  virtual int Add10MsAudio(const AudioFrame& audio_frame) OVERRIDE;
+
+  virtual const ReceiverInfo* GetReceiverInfo() const OVERRIDE;
+
+  virtual bool RegisterReceiveCodec(AudioDecoder* receive_codec) OVERRIDE;
+
+  virtual bool RegisterReceiveCodec(int decoder_type,
+                                    uint8_t payload_type) OVERRIDE;
+
+  virtual bool InsertPacket(const uint8_t* incoming_payload,
+                            int32_t payload_len_bytes,
+                            const WebRtcRTPHeader& rtp_info) OVERRIDE;
+
+  virtual bool InsertPayload(const uint8_t* incoming_payload,
+                             int32_t payload_len_byte,
+                             uint8_t payload_type,
+                             uint32_t timestamp) OVERRIDE;
+
+  virtual bool SetMinimumPlayoutDelay(int time_ms) OVERRIDE;
+
+  virtual bool SetMaximumPlayoutDelay(int time_ms) OVERRIDE;
+
+  virtual int LeastRequiredDelayMs() const OVERRIDE;
+
+  virtual bool PlayoutTimestamp(uint32_t* timestamp) OVERRIDE;
+
+  virtual bool Get10MsAudio(AudioFrame* audio_frame) OVERRIDE;
+
+  virtual bool NetworkStatistics(
+      ACMNetworkStatistics* network_statistics) OVERRIDE;
+
+  virtual bool EnableNack(size_t max_nack_list_size) OVERRIDE;
+
+  virtual void DisableNack() OVERRIDE;
+
+  virtual std::vector<uint16_t> GetNackList(
+      int round_trip_time_ms) const OVERRIDE;
+
+  virtual void GetDecodingCallStatistics(
+      AudioDecodingCallStats* call_stats) const OVERRIDE;
+
+ private:
+  // Temporary method to be used during redesign phase.
+  // Maps |codec_type| (a value from the anonymous enum in acm2::ACMCodecDB) to
+  // |codec_name|, |sample_rate_hz|, and |channels|.
+  // TODO(henrik.lundin) Remove this when no longer needed.
+  static bool MapCodecTypeToParameters(int codec_type,
+                                       std::string* codec_name,
+                                       int* sample_rate_hz,
+                                       int* channels);
+
+  scoped_ptr<acm2::AudioCodingModuleImpl> acm_old_;
+  int playout_frequency_hz_;
+};
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_