diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
new file mode 100644
index 0000000..fb3fe3e
--- /dev/null
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
@@ -0,0 +1,827 @@
+/*
+ *  Copyright (c) 2013 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/main/source/acm_receiver.h"
+
+#include <stdlib.h>  // malloc
+
+#include <algorithm>  // sort
+#include <vector>
+
+#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/main/source/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/main/source/acm_resampler.h"
+#include "webrtc/modules/audio_coding/main/source/nack.h"
+#include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h"
+#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
+#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/system_wrappers/interface/trace.h"
+
+namespace webrtc {
+
+namespace {
+
+const int kRtpHeaderSize = 12;
+const int kNeteqInitSampleRateHz = 16000;
+const int kNackThresholdPackets = 2;
+
+// |vad_activity_| field of |audio_frame| is set to |previous_audio_activity_|
+// before the call to this function.
+void SetAudioFrameActivityAndType(bool vad_enabled,
+                                  NetEqOutputType type,
+                                  AudioFrame* audio_frame) {
+  if (vad_enabled) {
+    switch (type) {
+      case kOutputNormal: {
+        audio_frame->vad_activity_ = AudioFrame::kVadActive;
+        audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
+        break;
+      }
+      case kOutputVADPassive: {
+        audio_frame->vad_activity_ = AudioFrame::kVadPassive;
+        audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
+        break;
+      }
+      case kOutputCNG: {
+        audio_frame->vad_activity_ = AudioFrame::kVadPassive;
+        audio_frame->speech_type_ = AudioFrame::kCNG;
+        break;
+      }
+      case kOutputPLC: {
+        // Don't change |audio_frame->vad_activity_|, it should be the same as
+        // |previous_audio_activity_|.
+        audio_frame->speech_type_ = AudioFrame::kPLC;
+        break;
+      }
+      case kOutputPLCtoCNG: {
+        audio_frame->vad_activity_ = AudioFrame::kVadPassive;
+        audio_frame->speech_type_ = AudioFrame::kPLCCNG;
+        break;
+      }
+      default:
+        assert(false);
+    }
+  } else {
+    // Always return kVadUnknown when receive VAD is inactive
+    audio_frame->vad_activity_ = AudioFrame::kVadUnknown;
+    switch (type) {
+      case kOutputNormal: {
+        audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
+        break;
+      }
+      case kOutputCNG: {
+        audio_frame->speech_type_ = AudioFrame::kCNG;
+        break;
+      }
+      case kOutputPLC: {
+        audio_frame->speech_type_ = AudioFrame::kPLC;
+        break;
+      }
+      case kOutputPLCtoCNG: {
+        audio_frame->speech_type_ = AudioFrame::kPLCCNG;
+        break;
+      }
+      case kOutputVADPassive: {
+        // Normally, we should no get any VAD decision if post-decoding VAD is
+        // not active. However, if post-decoding VAD has been active then
+        // disabled, we might be here for couple of frames.
+        audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
+        LOG_F(LS_WARNING) << "Post-decoding VAD is disabled but output is "
+            << "labeled VAD-passive";
+        break;
+      }
+      default:
+        assert(false);
+    }
+  }
+}
+
+// Is the given codec a CNG codec?
+bool IsCng(int codec_id) {
+  return (codec_id == ACMCodecDB::kCNNB || codec_id == ACMCodecDB::kCNWB ||
+      codec_id == ACMCodecDB::kCNSWB || codec_id == ACMCodecDB::kCNFB);
+}
+
+}  // namespace
+
+AcmReceiver::AcmReceiver()
+    : id_(0),
+      neteq_(NetEq::Create(kNeteqInitSampleRateHz)),
+      last_audio_decoder_(-1),  // Invalid value.
+      decode_lock_(RWLockWrapper::CreateRWLock()),
+      neteq_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
+      vad_enabled_(false),
+      previous_audio_activity_(AudioFrame::kVadUnknown),
+      current_sample_rate_hz_(kNeteqInitSampleRateHz),
+      nack_(),
+      nack_enabled_(false),
+      av_sync_(false),
+      initial_delay_manager_(),
+      missing_packets_sync_stream_(),
+      late_packets_sync_stream_() {
+  for (int n = 0; n < ACMCodecDB::kMaxNumCodecs; ++n) {
+    decoders_[n].registered = false;
+  }
+
+  // Make sure we are on the same page as NetEq, although the default behavior
+  // for NetEq has been VAD disabled.
+  if (vad_enabled_)
+    neteq_->EnableVad();
+  else
+    neteq_->DisableVad();
+}
+
+AcmReceiver::~AcmReceiver() {
+  delete neteq_;
+  delete decode_lock_;
+  delete neteq_crit_sect_;
+}
+
+int AcmReceiver::SetMinimumDelay(int delay_ms) {
+  if (neteq_->SetMinimumDelay(delay_ms))
+    return 0;
+  LOG_FERR1(LS_ERROR, "AcmReceiver::SetExtraDelay", delay_ms);
+  return -1;
+}
+
+int AcmReceiver::SetInitialDelay(int delay_ms) {
+  if (delay_ms < 0 || delay_ms > 10000) {
+    return -1;
+  }
+  CriticalSectionScoped lock(neteq_crit_sect_);
+
+  if (delay_ms == 0) {
+    av_sync_ = false;
+    initial_delay_manager_.reset();
+    missing_packets_sync_stream_.reset();
+    late_packets_sync_stream_.reset();
+    neteq_->SetMinimumDelay(0);
+    return 0;
+  }
+
+  if (av_sync_ && initial_delay_manager_->PacketBuffered()) {
+    // Too late for this API. Only works before a call is started.
+    return -1;
+  }
+
+  // Most of places NetEq calls are not within AcmReceiver's critical section to
+  // improve performance. Here, this call has to be placed before the following
+  // block, therefore, we keep it inside critical section. Otherwise, we have to
+  // release |neteq_crit_sect_| and acquire it again, which seems an overkill.
+  if (neteq_->SetMinimumDelay(delay_ms) < 0)
+    return -1;
+
+  const int kLatePacketThreshold = 5;
+  av_sync_ = true;
+  initial_delay_manager_.reset(new InitialDelayManager(delay_ms,
+                                                       kLatePacketThreshold));
+  missing_packets_sync_stream_.reset(new InitialDelayManager::SyncStream);
+  late_packets_sync_stream_.reset(new InitialDelayManager::SyncStream);
+  return 0;
+}
+
+int AcmReceiver::SetMaximumDelay(int delay_ms) {
+  if (neteq_->SetMaximumDelay(delay_ms))
+    return 0;
+  LOG_FERR1(LS_ERROR, "AcmReceiver::SetExtraDelay", delay_ms);
+  return -1;
+}
+
+int AcmReceiver::LeastRequiredDelayMs() const {
+  return neteq_->LeastRequiredDelayMs();
+}
+
+int AcmReceiver::current_sample_rate_hz() const {
+  CriticalSectionScoped lock(neteq_crit_sect_);
+  return current_sample_rate_hz_;
+}
+
+// TODO(turajs): use one set of enumerators, e.g. the one defined in
+// common_types.h
+void AcmReceiver::SetPlayoutMode(AudioPlayoutMode mode) {
+  enum NetEqPlayoutMode playout_mode = kPlayoutOn;
+  enum NetEqBackgroundNoiseMode bgn_mode = kBgnOn;
+  switch (mode) {
+    case voice:
+      playout_mode = kPlayoutOn;
+      bgn_mode = kBgnOn;
+      break;
+    case fax:  // No change to background noise mode.
+      playout_mode = kPlayoutFax;
+      bgn_mode = neteq_->BackgroundNoiseMode();
+      break;
+    case streaming:
+      playout_mode = kPlayoutStreaming;
+      bgn_mode = kBgnOff;
+      break;
+    case off:
+      playout_mode = kPlayoutOff;
+      bgn_mode = kBgnOff;
+      break;
+  }
+  neteq_->SetPlayoutMode(playout_mode);
+  neteq_->SetBackgroundNoiseMode(bgn_mode);
+}
+
+AudioPlayoutMode AcmReceiver::PlayoutMode() const {
+  AudioPlayoutMode acm_mode = voice;
+  NetEqPlayoutMode mode = neteq_->PlayoutMode();
+  switch (mode) {
+    case kPlayoutOn:
+      acm_mode = voice;
+      break;
+    case kPlayoutOff:
+      acm_mode = off;
+      break;
+    case kPlayoutFax:
+      acm_mode = fax;
+      break;
+    case kPlayoutStreaming:
+      acm_mode = streaming;
+      break;
+    default:
+      assert(false);
+  }
+  return acm_mode;
+}
+
+int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
+                              const uint8_t* incoming_payload,
+                              int length_payload) {
+  uint32_t receive_timestamp = 0;
+  InitialDelayManager::PacketType packet_type =
+      InitialDelayManager::kUndefinedPacket;
+  bool new_codec = false;
+  const RTPHeader* header = &rtp_header.header;  // Just a shorthand.
+
+  {
+    CriticalSectionScoped lock(neteq_crit_sect_);
+
+    int codec_id = RtpHeaderToCodecIndex(*header, incoming_payload);
+    if (codec_id < 0) {
+      LOG_F(LS_ERROR) << "Payload-type " << header->payloadType
+          << " is not registered.";
+      return -1;
+    }
+    assert(codec_id < ACMCodecDB::kMaxNumCodecs);
+    const int sample_rate_hz = ACMCodecDB::CodecFreq(codec_id);
+    receive_timestamp = NowInTimestamp(sample_rate_hz);
+
+    if (IsCng(codec_id)) {
+      // If this is a CNG while the audio codec is not mono skip pushing in
+      // packets into NetEq.
+      if (last_audio_decoder_ >= 0 &&
+        decoders_[last_audio_decoder_].channels > 1)
+        return 0;
+      packet_type = InitialDelayManager::kCngPacket;
+    } else if (codec_id == ACMCodecDB::kAVT) {
+      packet_type = InitialDelayManager::kAvtPacket;
+    } else {
+      if (codec_id != last_audio_decoder_) {
+        // This is either the first audio packet or send codec is changed.
+        // Therefore, either NetEq buffer is empty or will be flushed when this
+        // packet inserted. Note that |last_audio_decoder_| is initialized to
+        // an invalid value (-1), hence, the above condition is true for the
+        // very first audio packet.
+        new_codec = true;
+
+        // Updating NACK'sampling rate is required, either first packet is
+        // received or codec is changed. Furthermore, reset is required if codec
+        // is changed (NetEq flushes its buffer so NACK should reset its list).
+        if (nack_enabled_) {
+          assert(nack_.get());
+          nack_->Reset();
+          nack_->UpdateSampleRate(sample_rate_hz);
+        }
+        last_audio_decoder_ = codec_id;
+      }
+      packet_type = InitialDelayManager::kAudioPacket;
+    }
+
+    if (nack_enabled_) {
+      assert(nack_.get());
+      nack_->UpdateLastReceivedPacket(header->sequenceNumber,
+                                      header->timestamp);
+    }
+
+    if (av_sync_) {
+      assert(initial_delay_manager_.get());
+      assert(missing_packets_sync_stream_.get());
+      // This updates |initial_delay_manager_| and specifies an stream of
+      // sync-packets, if required to be inserted. We insert the sync-packets
+      // when AcmReceiver lock is released and |decoder_lock_| is acquired.
+      initial_delay_manager_->UpdateLastReceivedPacket(
+          rtp_header, receive_timestamp, packet_type, new_codec, sample_rate_hz,
+          missing_packets_sync_stream_.get());
+    }
+  }
+
+  {
+    WriteLockScoped lock_codecs(*decode_lock_);  // Lock to prevent an encoding.
+
+    // If |missing_packets_sync_stream_| is allocated then we are in AV-sync and
+    // we may need to insert sync-packets. We don't check |av_sync_| as we are
+    // outside AcmReceiver's critical section.
+    if (missing_packets_sync_stream_.get()) {
+      InsertStreamOfSyncPackets(missing_packets_sync_stream_.get());
+    }
+
+    if (neteq_->InsertPacket(rtp_header, incoming_payload, length_payload,
+                             receive_timestamp) < 0) {
+      LOG_FERR1(LS_ERROR, "AcmReceiver::InsertPacket", header->payloadType) <<
+          " Failed to insert packet";
+      return -1;
+    }
+  }
+  return 0;
+}
+
+int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
+  enum NetEqOutputType type;
+  int16_t* ptr_audio_buffer = audio_frame->data_;
+  int samples_per_channel;
+  int num_channels;
+  bool return_silence = false;
+
+  {
+    // Accessing members, take the lock.
+    CriticalSectionScoped lock(neteq_crit_sect_);
+
+    if (av_sync_) {
+      assert(initial_delay_manager_.get());
+      assert(late_packets_sync_stream_.get());
+      return_silence = GetSilence(desired_freq_hz, audio_frame);
+      uint32_t timestamp_now = NowInTimestamp(current_sample_rate_hz_);
+      initial_delay_manager_->LatePackets(timestamp_now,
+                                          late_packets_sync_stream_.get());
+    }
+
+    if (!return_silence) {
+      // This is our initial guess regarding whether a resampling will be
+      // required. It is based on previous sample rate of netEq. Most often,
+      // this is a correct guess, however, in case that incoming payload changes
+      // the resampling might might be needed. By doing so, we avoid an
+      // unnecessary memcpy().
+      if (desired_freq_hz != -1 &&
+          current_sample_rate_hz_ != desired_freq_hz) {
+        ptr_audio_buffer = audio_buffer_;
+      }
+    }
+  }
+
+  {
+    WriteLockScoped lock_codecs(*decode_lock_);  // Lock to prevent an encoding.
+
+    // If |late_packets_sync_stream_| is allocated then we have been in AV-sync
+    // mode and we might have to insert sync-packets.
+    if (late_packets_sync_stream_.get()) {
+      InsertStreamOfSyncPackets(late_packets_sync_stream_.get());
+      if (return_silence)  // Silence generated, don't pull from NetEq.
+        return 0;
+    }
+
+    if (neteq_->GetAudio(AudioFrame::kMaxDataSizeSamples,
+                         ptr_audio_buffer,
+                         &samples_per_channel,
+                         &num_channels, &type) != NetEq::kOK) {
+      LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio") << "NetEq Failed.";
+      return -1;
+    }
+  }
+
+  // Accessing members, take the lock.
+  CriticalSectionScoped lock(neteq_crit_sect_);
+
+  // Update NACK.
+  int decoded_sequence_num = 0;
+  uint32_t decoded_timestamp = 0;
+  bool update_nack = nack_enabled_ &&  // Update NACK only if it is enabled.
+      neteq_->DecodedRtpInfo(&decoded_sequence_num, &decoded_timestamp);
+  if (update_nack) {
+    assert(nack_.get());
+    nack_->UpdateLastDecodedPacket(decoded_sequence_num, decoded_timestamp);
+  }
+
+  // NetEq always returns 10 ms of audio.
+  current_sample_rate_hz_ = samples_per_channel * 100;
+
+  // Update if resampling is required.
+  bool need_resampling = (desired_freq_hz != -1) &&
+      (current_sample_rate_hz_ != desired_freq_hz);
+
+  if (ptr_audio_buffer == audio_buffer_) {
+    // Data is written to local buffer.
+    if (need_resampling) {
+      samples_per_channel = resampler_.Resample10Msec(
+          audio_buffer_, current_sample_rate_hz_, desired_freq_hz,
+          num_channels, audio_frame->data_);
+      if (samples_per_channel < 0) {
+        LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio") << "Resampler Failed.";
+        return -1;
+      }
+    } else {
+      // We might end up here ONLY if codec is changed.
+      memcpy(audio_frame->data_, audio_buffer_, samples_per_channel *
+             num_channels * sizeof(int16_t));
+    }
+  } else {
+    // Data is written into |audio_frame|.
+    if (need_resampling) {
+      // We might end up here ONLY if codec is changed.
+      samples_per_channel = resampler_.Resample10Msec(
+          audio_frame->data_, current_sample_rate_hz_, desired_freq_hz,
+          num_channels, audio_buffer_);
+      if (samples_per_channel < 0) {
+        LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio") << "Resampler Failed.";
+        return -1;
+      }
+      memcpy(audio_frame->data_, audio_buffer_, samples_per_channel *
+             num_channels * sizeof(int16_t));
+    }
+  }
+
+  audio_frame->num_channels_ = num_channels;
+  audio_frame->samples_per_channel_ = samples_per_channel;
+  audio_frame->sample_rate_hz_ = samples_per_channel * 100;
+
+  // Should set |vad_activity| before calling SetAudioFrameActivityAndType().
+  audio_frame->vad_activity_ = previous_audio_activity_;
+  SetAudioFrameActivityAndType(vad_enabled_, type, audio_frame);
+  previous_audio_activity_ = audio_frame->vad_activity_;
+  return 0;
+}
+
+int32_t AcmReceiver::AddCodec(int acm_codec_id,
+                              uint8_t payload_type,
+                              int channels,
+                              AudioDecoder* audio_decoder) {
+  assert(acm_codec_id >= 0 && acm_codec_id < ACMCodecDB::kMaxNumCodecs);
+  NetEqDecoder neteq_decoder = ACMCodecDB::neteq_decoders_[acm_codec_id];
+
+  CriticalSectionScoped lock(neteq_crit_sect_);
+
+  // The corresponding NetEq decoder ID.
+  // If this coder has been registered before.
+  if (decoders_[acm_codec_id].registered) {
+    if (decoders_[acm_codec_id].payload_type == payload_type) {
+      // Re-registering the same codec with the same payload-type. Do nothing
+      // and return.
+      return 0;
+    }
+
+    // Changing the payload-type of this codec. First unregister. Then register
+    // with new payload-type.
+    if (neteq_->RemovePayloadType(decoders_[acm_codec_id].payload_type) !=
+        NetEq::kOK) {
+      LOG_F(LS_ERROR) << "Cannot remover payload "
+          << decoders_[acm_codec_id].payload_type;
+      return -1;
+    }
+  }
+
+  int ret_val;
+  if (!audio_decoder) {
+    ret_val = neteq_->RegisterPayloadType(neteq_decoder, payload_type);
+  } else {
+    ret_val = neteq_->RegisterExternalDecoder(
+        audio_decoder, neteq_decoder,
+        ACMCodecDB::database_[acm_codec_id].plfreq, payload_type);
+  }
+  if (ret_val != NetEq::kOK) {
+    LOG_FERR3(LS_ERROR, "AcmReceiver::AddCodec", acm_codec_id, payload_type,
+              channels);
+    // Registration failed, delete the allocated space and set the pointer to
+    // NULL, for the record.
+    decoders_[acm_codec_id].registered = false;
+    return -1;
+  }
+
+  decoders_[acm_codec_id].registered = true;
+  decoders_[acm_codec_id].payload_type = payload_type;
+  decoders_[acm_codec_id].channels = channels;
+  return 0;
+}
+
+void AcmReceiver::EnableVad() {
+  neteq_->EnableVad();
+  CriticalSectionScoped lock(neteq_crit_sect_);
+  vad_enabled_ = true;
+}
+
+void AcmReceiver::DisableVad() {
+  neteq_->DisableVad();
+  CriticalSectionScoped lock(neteq_crit_sect_);
+  vad_enabled_ = false;
+}
+
+void AcmReceiver::FlushBuffers() {
+  neteq_->FlushBuffers();
+}
+
+// If failed in removing one of the codecs, this method continues to remove as
+// many as it can.
+int AcmReceiver::RemoveAllCodecs() {
+  int ret_val = 0;
+  CriticalSectionScoped lock(neteq_crit_sect_);
+  for (int n = 0; n < ACMCodecDB::kMaxNumCodecs; ++n) {
+    if (decoders_[n].registered) {
+      if (neteq_->RemovePayloadType(decoders_[n].payload_type) == 0) {
+        decoders_[n].registered = false;
+      } else {
+        LOG_F(LS_ERROR) << "Cannot remove payload "
+            << decoders_[n].payload_type;
+        ret_val = -1;
+      }
+    }
+  }
+  return ret_val;
+}
+
+int AcmReceiver::RemoveCodec(uint8_t payload_type) {
+  int codec_index = PayloadType2CodecIndex(payload_type);
+  if (codec_index < 0) {  // Such a payload-type is not registered.
+    LOG(LS_ERROR) << "payload_type " << payload_type << " is not registered"
+        " to be removed.";
+    return -1;
+  }
+  if (neteq_->RemovePayloadType(payload_type) != NetEq::kOK) {
+    LOG_FERR1(LS_ERROR, "AcmReceiver::RemoveCodec", payload_type);
+    return -1;
+  }
+  CriticalSectionScoped lock(neteq_crit_sect_);
+  decoders_[codec_index].registered = false;
+  return 0;
+}
+
+void AcmReceiver::set_id(int id) {
+  CriticalSectionScoped lock(neteq_crit_sect_);
+  id_ = id;
+}
+
+uint32_t AcmReceiver::PlayoutTimestamp() {
+  if (av_sync_) {
+    assert(initial_delay_manager_.get());
+    if (initial_delay_manager_->buffering())
+      return initial_delay_manager_->playout_timestamp();
+  }
+  return neteq_->PlayoutTimestamp();
+}
+
+int AcmReceiver::last_audio_codec_id() const {
+  CriticalSectionScoped lock(neteq_crit_sect_);
+  return last_audio_decoder_;
+}
+
+int AcmReceiver::last_audio_payload_type() const {
+  CriticalSectionScoped lock(neteq_crit_sect_);
+  if (last_audio_decoder_ < 0)
+    return -1;
+  assert(decoders_[last_audio_decoder_].registered);
+  return decoders_[last_audio_decoder_].payload_type;
+}
+
+int AcmReceiver::RedPayloadType() const {
+  CriticalSectionScoped lock(neteq_crit_sect_);
+  if (!decoders_[ACMCodecDB::kRED].registered) {
+    LOG_F(LS_WARNING) << "RED is not registered.";
+    return -1;
+  }
+  return decoders_[ACMCodecDB::kRED].payload_type;
+}
+
+int AcmReceiver::LastAudioCodec(CodecInst* codec) const {
+  CriticalSectionScoped lock(neteq_crit_sect_);
+  if (last_audio_decoder_ < 0) {
+    LOG_F(LS_WARNING) << "No audio payload is received, yet.";
+    return -1;
+  }
+  assert(decoders_[last_audio_decoder_].registered);
+  memcpy(codec, &ACMCodecDB::database_[last_audio_decoder_], sizeof(CodecInst));
+  codec->pltype = decoders_[last_audio_decoder_].payload_type;
+  codec->channels = decoders_[last_audio_decoder_].channels;
+  return 0;
+}
+
+void AcmReceiver::NetworkStatistics(ACMNetworkStatistics* acm_stat) {
+  NetEqNetworkStatistics neteq_stat;
+  // NetEq function always returns zero, so we don't check the return value.
+  neteq_->NetworkStatistics(&neteq_stat);
+
+  acm_stat->currentBufferSize = neteq_stat.current_buffer_size_ms;
+  acm_stat->preferredBufferSize = neteq_stat.preferred_buffer_size_ms;
+  acm_stat->jitterPeaksFound = neteq_stat.jitter_peaks_found;
+  acm_stat->currentPacketLossRate = neteq_stat.packet_loss_rate;
+  acm_stat->currentDiscardRate = neteq_stat.packet_discard_rate;
+  acm_stat->currentExpandRate = neteq_stat.expand_rate;
+  acm_stat->currentPreemptiveRate = neteq_stat.preemptive_rate;
+  acm_stat->currentAccelerateRate = neteq_stat.accelerate_rate;
+  acm_stat->clockDriftPPM = neteq_stat.clockdrift_ppm;
+
+  std::vector<int> waiting_times;
+  neteq_->WaitingTimes(&waiting_times);
+  size_t size = waiting_times.size();
+  if (size == 0) {
+    acm_stat->meanWaitingTimeMs = -1;
+    acm_stat->medianWaitingTimeMs = -1;
+    acm_stat->minWaitingTimeMs = -1;
+    acm_stat->maxWaitingTimeMs = -1;
+  } else {
+    std::sort(waiting_times.begin(), waiting_times.end());
+    if ((size & 0x1) == 0) {
+      acm_stat->medianWaitingTimeMs = (waiting_times[size / 2 - 1] +
+          waiting_times[size / 2]) / 2;
+    } else {
+      acm_stat->medianWaitingTimeMs = waiting_times[size / 2];
+    }
+    acm_stat->minWaitingTimeMs = waiting_times.front();
+    acm_stat->maxWaitingTimeMs = waiting_times.back();
+    double sum = 0;
+    for (size_t i = 0; i < size; ++i) {
+      sum += waiting_times[i];
+    }
+    acm_stat->meanWaitingTimeMs = static_cast<int>(sum / size);
+  }
+}
+
+int AcmReceiver::DecoderByPayloadType(uint8_t payload_type,
+                                      CodecInst* codec) const {
+  CriticalSectionScoped lock(neteq_crit_sect_);
+  int codec_index = PayloadType2CodecIndex(payload_type);
+  if (codec_index < 0) {
+    LOG_FERR1(LS_ERROR, "AcmReceiver::DecoderByPayloadType", payload_type);
+    return -1;
+  }
+  memcpy(codec, &ACMCodecDB::database_[codec_index], sizeof(CodecInst));
+  codec->pltype = decoders_[codec_index].payload_type;
+  codec->channels = decoders_[codec_index].channels;
+  return 0;
+}
+
+int AcmReceiver::PayloadType2CodecIndex(uint8_t payload_type) const {
+  for (int n = 0; n < ACMCodecDB::kMaxNumCodecs; ++n) {
+    if (decoders_[n].registered && decoders_[n].payload_type == payload_type) {
+      return n;
+    }
+  }
+  return -1;
+}
+
+int AcmReceiver::EnableNack(size_t max_nack_list_size) {
+  // Don't do anything if |max_nack_list_size| is out of range.
+  if (max_nack_list_size == 0 || max_nack_list_size > Nack::kNackListSizeLimit)
+    return -1;
+
+  CriticalSectionScoped lock(neteq_crit_sect_);
+  if (!nack_enabled_) {
+    nack_.reset(Nack::Create(kNackThresholdPackets));
+    nack_enabled_ = true;
+
+    // Sampling rate might need to be updated if we change from disable to
+    // enable. Do it if the receive codec is valid.
+    if (last_audio_decoder_ >= 0) {
+      nack_->UpdateSampleRate(
+          ACMCodecDB::database_[last_audio_decoder_].plfreq);
+    }
+  }
+  return nack_->SetMaxNackListSize(max_nack_list_size);
+}
+
+void AcmReceiver::DisableNack() {
+  CriticalSectionScoped lock(neteq_crit_sect_);
+  nack_.reset();  // Memory is released.
+  nack_enabled_ = false;
+}
+
+std::vector<uint16_t> AcmReceiver::GetNackList(
+    int round_trip_time_ms) const {
+  CriticalSectionScoped lock(neteq_crit_sect_);
+  if (round_trip_time_ms < 0) {
+    WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_,
+                 "GetNackList: round trip time cannot be negative."
+                 " round_trip_time_ms=%d", round_trip_time_ms);
+  }
+  if (nack_enabled_ && round_trip_time_ms >= 0) {
+    assert(nack_.get());
+    return nack_->GetNackList(round_trip_time_ms);
+  }
+  std::vector<uint16_t> empty_list;
+  return empty_list;
+}
+
+void AcmReceiver::ResetInitialDelay() {
+  {
+    CriticalSectionScoped lock(neteq_crit_sect_);
+    av_sync_ = false;
+    initial_delay_manager_.reset(NULL);
+    missing_packets_sync_stream_.reset(NULL);
+    late_packets_sync_stream_.reset(NULL);
+  }
+  neteq_->SetMinimumDelay(0);
+  // TODO(turajs): Should NetEq Buffer be flushed?
+}
+
+// This function is called within critical section, no need to acquire a lock.
+bool AcmReceiver::GetSilence(int desired_sample_rate_hz, AudioFrame* frame) {
+  assert(av_sync_);
+  assert(initial_delay_manager_.get());
+  if (!initial_delay_manager_->buffering()) {
+    return false;
+  }
+
+  // We stop accumulating packets, if the number of packets or the total size
+  // exceeds a threshold.
+  int num_packets;
+  int max_num_packets;
+  int buffer_size_byte;
+  int max_buffer_size_byte;
+  const float kBufferingThresholdScale = 0.9;
+  neteq_->PacketBufferStatistics(&num_packets, &max_num_packets,
+                                 &buffer_size_byte, &max_buffer_size_byte);
+  if (num_packets > max_num_packets * kBufferingThresholdScale ||
+      buffer_size_byte > max_buffer_size_byte * kBufferingThresholdScale) {
+    initial_delay_manager_->DisableBuffering();
+    return false;
+  }
+
+  // Set the values if already got a packet, otherwise set to default values.
+  if (last_audio_decoder_ >= 0) {
+    current_sample_rate_hz_ = ACMCodecDB::database_[last_audio_decoder_].plfreq;
+    frame->num_channels_ = decoders_[last_audio_decoder_].channels;
+  } else {
+    current_sample_rate_hz_ = kNeteqInitSampleRateHz;
+    frame->num_channels_ = 1;
+  }
+
+  // Set the audio frame's sampling frequency.
+  if (desired_sample_rate_hz > 0) {
+    frame->sample_rate_hz_ = desired_sample_rate_hz;
+  } else {
+    frame->sample_rate_hz_ = current_sample_rate_hz_;
+  }
+
+  frame->samples_per_channel_ = frame->sample_rate_hz_ / 100;  // Always 10 ms.
+  frame->speech_type_ = AudioFrame::kCNG;
+  frame->vad_activity_ = AudioFrame::kVadPassive;
+  frame->energy_ = 0;
+  int samples = frame->samples_per_channel_ * frame->num_channels_;
+  memset(frame->data_, 0, samples * sizeof(int16_t));
+  return true;
+}
+
+NetEqBackgroundNoiseMode AcmReceiver::BackgroundNoiseModeForTest() const {
+  return neteq_->BackgroundNoiseMode();
+}
+
+int AcmReceiver::RtpHeaderToCodecIndex(
+    const RTPHeader &rtp_header, const uint8_t* payload) const {
+  uint8_t payload_type = rtp_header.payloadType;
+  if (decoders_[ACMCodecDB::kRED].registered &&
+      payload_type == decoders_[ACMCodecDB::kRED].payload_type) {
+    // This is a RED packet, get the payload of the audio codec.
+    payload_type = payload[0] & 0x7F;
+  }
+
+  // Check if the payload is registered.
+  return PayloadType2CodecIndex(payload_type);
+}
+
+uint32_t AcmReceiver::NowInTimestamp(int decoder_sampling_rate) const {
+  // Down-cast the time to (32-6)-bit since we only care about
+  // the least significant bits. (32-6) bits cover 2^(32-6) = 67108864 ms.
+  // We masked 6 most significant bits of 32-bit so there is no overflow in
+  // the conversion from milliseconds to timestamp.
+  const uint32_t now_in_ms = static_cast<uint32_t>(
+      TickTime::MillisecondTimestamp() & 0x03ffffff);
+  return static_cast<uint32_t>(
+      (decoder_sampling_rate / 1000) * now_in_ms);
+}
+
+// This function only interacts with |neteq_|, therefore, it does not have to
+// be within critical section of AcmReceiver. It is inserting packets
+// into NetEq, so we call it when |decode_lock_| is acquired. However, this is
+// not essential as sync-packets do not interact with codecs (especially BWE).
+void AcmReceiver::InsertStreamOfSyncPackets(
+    InitialDelayManager::SyncStream* sync_stream) {
+  assert(sync_stream);
+  assert(av_sync_);
+  for (int n = 0; n < sync_stream->num_sync_packets; ++n) {
+    neteq_->InsertSyncPacket(sync_stream->rtp_info,
+                             sync_stream->receive_timestamp);
+    ++sync_stream->rtp_info.header.sequenceNumber;
+    sync_stream->rtp_info.header.timestamp += sync_stream->timestamp_step;
+    sync_stream->receive_timestamp += sync_stream->timestamp_step;
+  }
+}
+
+}  // namespace webrtc