/*
 *  Copyright (c) 2012 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/acm2/audio_coding_module_impl.h"

#include <assert.h>
#include <stdlib.h>
#include <vector>

#include "webrtc/base/checks.h"
#include "webrtc/base/safe_conversions.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_common_defs.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/logging.h"
#include "webrtc/system_wrappers/include/metrics.h"
#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/typedefs.h"

namespace webrtc {

namespace acm2 {

namespace {

// 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,
            int16_t* out_buff) {
  if (length_out_buff < frame.samples_per_channel_) {
    return -1;
  }
  for (size_t n = 0; n < frame.samples_per_channel_; ++n)
    out_buff[n] = (frame.data_[2 * n] + frame.data_[2 * n + 1]) >> 1;
  return 0;
}

// Mono-to-stereo can be used as in-place.
int UpMix(const AudioFrame& frame, size_t length_out_buff, int16_t* out_buff) {
  if (length_out_buff < frame.samples_per_channel_) {
    return -1;
  }
  for (size_t n = frame.samples_per_channel_; n != 0; --n) {
    size_t i = n - 1;
    int16_t sample = frame.data_[i];
    out_buff[2 * i + 1] = sample;
    out_buff[2 * i] = sample;
  }
  return 0;
}

void ConvertEncodedInfoToFragmentationHeader(
    const AudioEncoder::EncodedInfo& info,
    RTPFragmentationHeader* frag) {
  if (info.redundant.empty()) {
    frag->fragmentationVectorSize = 0;
    return;
  }

  frag->VerifyAndAllocateFragmentationHeader(
      static_cast<uint16_t>(info.redundant.size()));
  frag->fragmentationVectorSize = static_cast<uint16_t>(info.redundant.size());
  size_t offset = 0;
  for (size_t i = 0; i < info.redundant.size(); ++i) {
    frag->fragmentationOffset[i] = offset;
    offset += info.redundant[i].encoded_bytes;
    frag->fragmentationLength[i] = info.redundant[i].encoded_bytes;
    frag->fragmentationTimeDiff[i] = rtc::checked_cast<uint16_t>(
        info.encoded_timestamp - info.redundant[i].encoded_timestamp);
    frag->fragmentationPlType[i] = info.redundant[i].payload_type;
  }
}
}  // namespace

void AudioCodingModuleImpl::ChangeLogger::MaybeLog(int value) {
  if (value != last_value_ || first_time_) {
    first_time_ = false;
    last_value_ = value;
    RTC_HISTOGRAM_COUNTS_100(histogram_name_, value);
  }
}

AudioCodingModuleImpl::AudioCodingModuleImpl(
    const AudioCodingModule::Config& config)
    : acm_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
      id_(config.id),
      expected_codec_ts_(0xD87F3F9F),
      expected_in_ts_(0xD87F3F9F),
      receiver_(config),
      bitrate_logger_("WebRTC.Audio.TargetBitrateInKbps"),
      previous_pltype_(255),
      receiver_initialized_(false),
      first_10ms_data_(false),
      first_frame_(true),
      callback_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
      packetization_callback_(NULL),
      vad_callback_(NULL) {
  if (InitializeReceiverSafe() < 0) {
    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                 "Cannot initialize receiver");
  }
  WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceAudioCoding, id_, "Created");
}

AudioCodingModuleImpl::~AudioCodingModuleImpl() = default;

int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
  AudioEncoder::EncodedInfo encoded_info;
  uint8_t previous_pltype;

  // Check if there is an encoder before.
  if (!HaveValidEncoder("Process"))
    return -1;

  AudioEncoder* audio_encoder = codec_manager_.CurrentEncoder();
  // Scale the timestamp to the codec's RTP timestamp rate.
  uint32_t rtp_timestamp =
      first_frame_ ? input_data.input_timestamp
                   : last_rtp_timestamp_ +
                         rtc::CheckedDivExact(
                             input_data.input_timestamp - last_timestamp_,
                             static_cast<uint32_t>(rtc::CheckedDivExact(
                                 audio_encoder->SampleRateHz(),
                                 audio_encoder->RtpTimestampRateHz())));
  last_timestamp_ = input_data.input_timestamp;
  last_rtp_timestamp_ = rtp_timestamp;
  first_frame_ = false;

  encode_buffer_.SetSize(audio_encoder->MaxEncodedBytes());
  encoded_info = audio_encoder->Encode(
      rtp_timestamp, input_data.audio, input_data.length_per_channel,
      encode_buffer_.size(), encode_buffer_.data());
  encode_buffer_.SetSize(encoded_info.encoded_bytes);
  bitrate_logger_.MaybeLog(audio_encoder->GetTargetBitrate() / 1000);
  if (encode_buffer_.size() == 0 && !encoded_info.send_even_if_empty) {
    // Not enough data.
    return 0;
  }
  previous_pltype = previous_pltype_;  // Read it while we have the critsect.

  RTPFragmentationHeader my_fragmentation;
  ConvertEncodedInfoToFragmentationHeader(encoded_info, &my_fragmentation);
  FrameType frame_type;
  if (encode_buffer_.size() == 0 && encoded_info.send_even_if_empty) {
    frame_type = kEmptyFrame;
    encoded_info.payload_type = previous_pltype;
  } else {
    RTC_DCHECK_GT(encode_buffer_.size(), 0u);
    frame_type = encoded_info.speech ? kAudioFrameSpeech : kAudioFrameCN;
  }

  {
    CriticalSectionScoped lock(callback_crit_sect_.get());
    if (packetization_callback_) {
      packetization_callback_->SendData(
          frame_type, encoded_info.payload_type, encoded_info.encoded_timestamp,
          encode_buffer_.data(), encode_buffer_.size(),
          my_fragmentation.fragmentationVectorSize > 0 ? &my_fragmentation
                                                       : nullptr);
    }

    if (vad_callback_) {
      // Callback with VAD decision.
      vad_callback_->InFrameType(frame_type);
    }
  }
  previous_pltype_ = encoded_info.payload_type;
  return static_cast<int32_t>(encode_buffer_.size());
}

/////////////////////////////////////////
//   Sender
//

// Can be called multiple times for Codec, CNG, RED.
int AudioCodingModuleImpl::RegisterSendCodec(const CodecInst& send_codec) {
  CriticalSectionScoped lock(acm_crit_sect_.get());
  return codec_manager_.RegisterEncoder(send_codec);
}

void AudioCodingModuleImpl::RegisterExternalSendCodec(
    AudioEncoder* external_speech_encoder) {
  CriticalSectionScoped lock(acm_crit_sect_.get());
  codec_manager_.RegisterEncoder(external_speech_encoder);
}

// Get current send codec.
int AudioCodingModuleImpl::SendCodec(CodecInst* current_codec) const {
  CriticalSectionScoped lock(acm_crit_sect_.get());
  return codec_manager_.GetCodecInst(current_codec);
}

// Get current send frequency.
int AudioCodingModuleImpl::SendFrequency() const {
  WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
               "SendFrequency()");
  CriticalSectionScoped lock(acm_crit_sect_.get());

  if (!codec_manager_.CurrentEncoder()) {
    WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
                 "SendFrequency Failed, no codec is registered");
    return -1;
  }

  return codec_manager_.CurrentEncoder()->SampleRateHz();
}

void AudioCodingModuleImpl::SetBitRate(int bitrate_bps) {
  CriticalSectionScoped lock(acm_crit_sect_.get());
  if (codec_manager_.CurrentEncoder()) {
    codec_manager_.CurrentEncoder()->SetTargetBitrate(bitrate_bps);
  }
}

// Register a transport callback which will be called to deliver
// the encoded buffers.
int AudioCodingModuleImpl::RegisterTransportCallback(
    AudioPacketizationCallback* transport) {
  CriticalSectionScoped lock(callback_crit_sect_.get());
  packetization_callback_ = transport;
  return 0;
}

// Add 10MS of raw (PCM) audio data to the encoder.
int AudioCodingModuleImpl::Add10MsData(const AudioFrame& audio_frame) {
  InputData input_data;
  CriticalSectionScoped lock(acm_crit_sect_.get());
  int r = Add10MsDataInternal(audio_frame, &input_data);
  return r < 0 ? r : Encode(input_data);
}

int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame,
                                               InputData* input_data) {
  if (audio_frame.samples_per_channel_ == 0) {
    assert(false);
    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                 "Cannot Add 10 ms audio, payload length is zero");
    return -1;
  }

  if (audio_frame.sample_rate_hz_ > 48000) {
    assert(false);
    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                 "Cannot Add 10 ms audio, input frequency not valid");
    return -1;
  }

  // If the length and frequency matches. We currently just support raw PCM.
  if (static_cast<size_t>(audio_frame.sample_rate_hz_ / 100) !=
      audio_frame.samples_per_channel_) {
    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                 "Cannot Add 10 ms audio, input frequency and length doesn't"
                 " match");
    return -1;
  }

  if (audio_frame.num_channels_ != 1 && audio_frame.num_channels_ != 2) {
    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                 "Cannot Add 10 ms audio, invalid number of channels.");
    return -1;
  }

  // Do we have a codec registered?
  if (!HaveValidEncoder("Add10MsData")) {
    return -1;
  }

  const AudioFrame* ptr_frame;
  // Perform a resampling, also down-mix if it is required and can be
  // performed before resampling (a down mix prior to resampling will take
  // place if both primary and secondary encoders are mono and input is in
  // stereo).
  if (PreprocessToAddData(audio_frame, &ptr_frame) < 0) {
    return -1;
  }

  // Check whether we need an up-mix or down-mix?
  bool remix = ptr_frame->num_channels_ !=
               codec_manager_.CurrentEncoder()->NumChannels();

  if (remix) {
    if (ptr_frame->num_channels_ == 1) {
      if (UpMix(*ptr_frame, WEBRTC_10MS_PCM_AUDIO, input_data->buffer) < 0)
        return -1;
    } else {
      if (DownMix(*ptr_frame, WEBRTC_10MS_PCM_AUDIO, input_data->buffer) < 0)
        return -1;
    }
  }

  // When adding data to encoders this pointer is pointing to an audio buffer
  // with correct number of channels.
  const int16_t* ptr_audio = ptr_frame->data_;

  // For pushing data to primary, point the |ptr_audio| to correct buffer.
  if (codec_manager_.CurrentEncoder()->NumChannels() !=
      ptr_frame->num_channels_)
    ptr_audio = input_data->buffer;

  input_data->input_timestamp = ptr_frame->timestamp_;
  input_data->audio = ptr_audio;
  input_data->length_per_channel = ptr_frame->samples_per_channel_;
  input_data->audio_channel = codec_manager_.CurrentEncoder()->NumChannels();

  return 0;
}

// Perform a resampling and down-mix if required. We down-mix only if
// encoder is mono and input is stereo. In case of dual-streaming, both
// encoders has to be mono for down-mix to take place.
// |*ptr_out| will point to the pre-processed audio-frame. If no pre-processing
// is required, |*ptr_out| points to |in_frame|.
int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
                                               const AudioFrame** ptr_out) {
  bool resample = (in_frame.sample_rate_hz_ !=
                   codec_manager_.CurrentEncoder()->SampleRateHz());

  // This variable is true if primary codec and secondary codec (if exists)
  // are both mono and input is stereo.
  bool down_mix = (in_frame.num_channels_ == 2) &&
                  (codec_manager_.CurrentEncoder()->NumChannels() == 1);

  if (!first_10ms_data_) {
    expected_in_ts_ = in_frame.timestamp_;
    expected_codec_ts_ = in_frame.timestamp_;
    first_10ms_data_ = true;
  } else if (in_frame.timestamp_ != expected_in_ts_) {
    // TODO(turajs): Do we need a warning here.
    expected_codec_ts_ +=
        (in_frame.timestamp_ - expected_in_ts_) *
        static_cast<uint32_t>(
            (static_cast<double>(
                 codec_manager_.CurrentEncoder()->SampleRateHz()) /
             static_cast<double>(in_frame.sample_rate_hz_)));
    expected_in_ts_ = in_frame.timestamp_;
  }


  if (!down_mix && !resample) {
    // No pre-processing is required.
    expected_in_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
    expected_codec_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
    *ptr_out = &in_frame;
    return 0;
  }

  *ptr_out = &preprocess_frame_;
  preprocess_frame_.num_channels_ = in_frame.num_channels_;
  int16_t audio[WEBRTC_10MS_PCM_AUDIO];
  const int16_t* src_ptr_audio = in_frame.data_;
  int16_t* dest_ptr_audio = preprocess_frame_.data_;
  if (down_mix) {
    // If a resampling is required the output of a down-mix is written into a
    // local buffer, otherwise, it will be written to the output frame.
    if (resample)
      dest_ptr_audio = audio;
    if (DownMix(in_frame, WEBRTC_10MS_PCM_AUDIO, dest_ptr_audio) < 0)
      return -1;
    preprocess_frame_.num_channels_ = 1;
    // Set the input of the resampler is the down-mixed signal.
    src_ptr_audio = audio;
  }

  preprocess_frame_.timestamp_ = expected_codec_ts_;
  preprocess_frame_.samples_per_channel_ = in_frame.samples_per_channel_;
  preprocess_frame_.sample_rate_hz_ = in_frame.sample_rate_hz_;
  // If it is required, we have to do a resampling.
  if (resample) {
    // The result of the resampler is written to output frame.
    dest_ptr_audio = preprocess_frame_.data_;

    int samples_per_channel = resampler_.Resample10Msec(
        src_ptr_audio, in_frame.sample_rate_hz_,
        codec_manager_.CurrentEncoder()->SampleRateHz(),
        preprocess_frame_.num_channels_, AudioFrame::kMaxDataSizeSamples,
        dest_ptr_audio);

    if (samples_per_channel < 0) {
      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                   "Cannot add 10 ms audio, resampling failed");
      return -1;
    }
    preprocess_frame_.samples_per_channel_ =
        static_cast<size_t>(samples_per_channel);
    preprocess_frame_.sample_rate_hz_ =
        codec_manager_.CurrentEncoder()->SampleRateHz();
  }

  expected_codec_ts_ +=
      static_cast<uint32_t>(preprocess_frame_.samples_per_channel_);
  expected_in_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);

  return 0;
}

/////////////////////////////////////////
//   (RED) Redundant Coding
//

bool AudioCodingModuleImpl::REDStatus() const {
  CriticalSectionScoped lock(acm_crit_sect_.get());
  return codec_manager_.red_enabled();
}

// Configure RED status i.e on/off.
int AudioCodingModuleImpl::SetREDStatus(
#ifdef WEBRTC_CODEC_RED
    bool enable_red) {
  CriticalSectionScoped lock(acm_crit_sect_.get());
  return codec_manager_.SetCopyRed(enable_red) ? 0 : -1;
#else
    bool /* enable_red */) {
  WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_,
               "  WEBRTC_CODEC_RED is undefined");
  return -1;
#endif
}

/////////////////////////////////////////
//   (FEC) Forward Error Correction (codec internal)
//

bool AudioCodingModuleImpl::CodecFEC() const {
  CriticalSectionScoped lock(acm_crit_sect_.get());
  return codec_manager_.codec_fec_enabled();
}

int AudioCodingModuleImpl::SetCodecFEC(bool enable_codec_fec) {
  CriticalSectionScoped lock(acm_crit_sect_.get());
  return codec_manager_.SetCodecFEC(enable_codec_fec);
}

int AudioCodingModuleImpl::SetPacketLossRate(int loss_rate) {
  CriticalSectionScoped lock(acm_crit_sect_.get());
  if (HaveValidEncoder("SetPacketLossRate")) {
    codec_manager_.CurrentEncoder()->SetProjectedPacketLossRate(loss_rate /
                                                                100.0);
  }
  return 0;
}

/////////////////////////////////////////
//   (VAD) Voice Activity Detection
//
int AudioCodingModuleImpl::SetVAD(bool enable_dtx,
                                  bool enable_vad,
                                  ACMVADMode mode) {
  // Note: |enable_vad| is not used; VAD is enabled based on the DTX setting.
  RTC_DCHECK_EQ(enable_dtx, enable_vad);
  CriticalSectionScoped lock(acm_crit_sect_.get());
  return codec_manager_.SetVAD(enable_dtx, mode);
}

// Get VAD/DTX settings.
int AudioCodingModuleImpl::VAD(bool* dtx_enabled, bool* vad_enabled,
                               ACMVADMode* mode) const {
  CriticalSectionScoped lock(acm_crit_sect_.get());
  codec_manager_.VAD(dtx_enabled, vad_enabled, mode);
  return 0;
}

/////////////////////////////////////////
//   Receiver
//

int AudioCodingModuleImpl::InitializeReceiver() {
  CriticalSectionScoped lock(acm_crit_sect_.get());
  return InitializeReceiverSafe();
}

// Initialize receiver, resets codec database etc.
int AudioCodingModuleImpl::InitializeReceiverSafe() {
  // If the receiver is already initialized then we want to destroy any
  // existing decoders. After a call to this function, we should have a clean
  // start-up.
  if (receiver_initialized_) {
    if (receiver_.RemoveAllCodecs() < 0)
      return -1;
  }
  receiver_.set_id(id_);
  receiver_.ResetInitialDelay();
  receiver_.SetMinimumDelay(0);
  receiver_.SetMaximumDelay(0);
  receiver_.FlushBuffers();

  // Register RED and CN.
  auto db = 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) < 0) {
        WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                     "Cannot register master codec.");
        return -1;
      }
    }
  }
  receiver_initialized_ = true;
  return 0;
}

// Get current receive frequency.
int AudioCodingModuleImpl::ReceiveFrequency() const {
  WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
               "ReceiveFrequency()");

  CriticalSectionScoped lock(acm_crit_sect_.get());

  int codec_id = receiver_.last_audio_codec_id();

  return codec_id < 0 ? receiver_.current_sample_rate_hz() :
                        ACMCodecDB::database_[codec_id].plfreq;
}

// Get current playout frequency.
int AudioCodingModuleImpl::PlayoutFrequency() const {
  WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
               "PlayoutFrequency()");

  CriticalSectionScoped lock(acm_crit_sect_.get());

  return receiver_.current_sample_rate_hz();
}

// Register possible receive codecs, can be called multiple times,
// for codecs, CNG (NB, WB and SWB), DTMF, RED.
int AudioCodingModuleImpl::RegisterReceiveCodec(const CodecInst& codec) {
  CriticalSectionScoped lock(acm_crit_sect_.get());
  RTC_DCHECK(receiver_initialized_);
  if (codec.channels > 2 || codec.channels < 0) {
    LOG_F(LS_ERROR) << "Unsupported number of channels: " << codec.channels;
    return -1;
  }

  auto codec_id =
      RentACodec::CodecIdByParams(codec.plname, codec.plfreq, codec.channels);
  if (!codec_id) {
    LOG_F(LS_ERROR) << "Wrong codec params to be registered as receive codec";
    return -1;
  }
  auto codec_index = RentACodec::CodecIndexFromId(*codec_id);
  RTC_CHECK(codec_index) << "Invalid codec ID: " << static_cast<int>(*codec_id);

  // Check if the payload-type is valid.
  if (!ACMCodecDB::ValidPayloadType(codec.pltype)) {
    LOG_F(LS_ERROR) << "Invalid payload type " << codec.pltype << " for "
                    << codec.plname;
    return -1;
  }

  // Get |decoder| associated with |codec|. |decoder| is NULL if |codec| does
  // not own its decoder.
  return receiver_.AddCodec(*codec_index, codec.pltype, codec.channels,
                            codec.plfreq,
                            codec_manager_.GetAudioDecoder(codec));
}

int AudioCodingModuleImpl::RegisterExternalReceiveCodec(
    int rtp_payload_type,
    AudioDecoder* external_decoder,
    int sample_rate_hz,
    int num_channels) {
  CriticalSectionScoped lock(acm_crit_sect_.get());
  RTC_DCHECK(receiver_initialized_);
  if (num_channels > 2 || num_channels < 0) {
    LOG_F(LS_ERROR) << "Unsupported number of channels: " << num_channels;
    return -1;
  }

  // Check if the payload-type is valid.
  if (!ACMCodecDB::ValidPayloadType(rtp_payload_type)) {
    LOG_F(LS_ERROR) << "Invalid payload-type " << rtp_payload_type
                    << " for external decoder.";
    return -1;
  }

  return receiver_.AddCodec(-1 /* external */, rtp_payload_type, num_channels,
                            sample_rate_hz, external_decoder);
}

// Get current received codec.
int AudioCodingModuleImpl::ReceiveCodec(CodecInst* current_codec) const {
  CriticalSectionScoped lock(acm_crit_sect_.get());
  return receiver_.LastAudioCodec(current_codec);
}

// Incoming packet from network parsed and ready for decode.
int AudioCodingModuleImpl::IncomingPacket(const uint8_t* incoming_payload,
                                          const size_t payload_length,
                                          const WebRtcRTPHeader& rtp_header) {
  return receiver_.InsertPacket(rtp_header, incoming_payload, payload_length);
}

// Minimum playout delay (Used for lip-sync).
int AudioCodingModuleImpl::SetMinimumPlayoutDelay(int time_ms) {
  if ((time_ms < 0) || (time_ms > 10000)) {
    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                 "Delay must be in the range of 0-1000 milliseconds.");
    return -1;
  }
  return receiver_.SetMinimumDelay(time_ms);
}

int AudioCodingModuleImpl::SetMaximumPlayoutDelay(int time_ms) {
  if ((time_ms < 0) || (time_ms > 10000)) {
    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                 "Delay must be in the range of 0-1000 milliseconds.");
    return -1;
  }
  return receiver_.SetMaximumDelay(time_ms);
}

// Get 10 milliseconds of raw audio data to play out.
// Automatic resample to the requested frequency.
int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
                                           AudioFrame* audio_frame) {
  // GetAudio always returns 10 ms, at the requested sample rate.
  if (receiver_.GetAudio(desired_freq_hz, audio_frame) != 0) {
    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                 "PlayoutData failed, RecOut Failed");
    return -1;
  }
  audio_frame->id_ = id_;
  return 0;
}

/////////////////////////////////////////
//   Statistics
//

// TODO(turajs) change the return value to void. Also change the corresponding
// NetEq function.
int AudioCodingModuleImpl::GetNetworkStatistics(NetworkStatistics* statistics) {
  receiver_.GetNetworkStatistics(statistics);
  return 0;
}

int AudioCodingModuleImpl::RegisterVADCallback(ACMVADCallback* vad_callback) {
  WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, id_,
               "RegisterVADCallback()");
  CriticalSectionScoped lock(callback_crit_sect_.get());
  vad_callback_ = vad_callback;
  return 0;
}

// TODO(kwiberg): Remove this method, and have callers call IncomingPacket
// instead. The translation logic and state belong with them, not with
// AudioCodingModuleImpl.
int AudioCodingModuleImpl::IncomingPayload(const uint8_t* incoming_payload,
                                           size_t payload_length,
                                           uint8_t payload_type,
                                           uint32_t timestamp) {
  // We are not acquiring any lock when interacting with |aux_rtp_header_| no
  // other method uses this member variable.
  if (!aux_rtp_header_) {
    // This is the first time that we are using |dummy_rtp_header_|
    // so we have to create it.
    aux_rtp_header_.reset(new WebRtcRTPHeader);
    aux_rtp_header_->header.payloadType = payload_type;
    // Don't matter in this case.
    aux_rtp_header_->header.ssrc = 0;
    aux_rtp_header_->header.markerBit = false;
    // Start with random numbers.
    aux_rtp_header_->header.sequenceNumber = 0x1234;  // Arbitrary.
    aux_rtp_header_->type.Audio.channel = 1;
  }

  aux_rtp_header_->header.timestamp = timestamp;
  IncomingPacket(incoming_payload, payload_length, *aux_rtp_header_);
  // Get ready for the next payload.
  aux_rtp_header_->header.sequenceNumber++;
  return 0;
}

int AudioCodingModuleImpl::SetOpusApplication(OpusApplicationMode application) {
  CriticalSectionScoped lock(acm_crit_sect_.get());
  if (!HaveValidEncoder("SetOpusApplication")) {
    return -1;
  }
  if (!codec_manager_.CurrentEncoderIsOpus())
    return -1;
  AudioEncoder::Application app;
  switch (application) {
    case kVoip:
      app = AudioEncoder::Application::kSpeech;
      break;
    case kAudio:
      app = AudioEncoder::Application::kAudio;
      break;
    default:
      FATAL();
      return 0;
  }
  return codec_manager_.CurrentEncoder()->SetApplication(app) ? 0 : -1;
}

// Informs Opus encoder of the maximum playback rate the receiver will render.
int AudioCodingModuleImpl::SetOpusMaxPlaybackRate(int frequency_hz) {
  CriticalSectionScoped lock(acm_crit_sect_.get());
  if (!HaveValidEncoder("SetOpusMaxPlaybackRate")) {
    return -1;
  }
  if (!codec_manager_.CurrentEncoderIsOpus())
    return -1;
  codec_manager_.CurrentEncoder()->SetMaxPlaybackRate(frequency_hz);
  return 0;
}

int AudioCodingModuleImpl::EnableOpusDtx() {
  CriticalSectionScoped lock(acm_crit_sect_.get());
  if (!HaveValidEncoder("EnableOpusDtx")) {
    return -1;
  }
  if (!codec_manager_.CurrentEncoderIsOpus())
    return -1;
  return codec_manager_.CurrentEncoder()->SetDtx(true) ? 0 : -1;
}

int AudioCodingModuleImpl::DisableOpusDtx() {
  CriticalSectionScoped lock(acm_crit_sect_.get());
  if (!HaveValidEncoder("DisableOpusDtx")) {
    return -1;
  }
  if (!codec_manager_.CurrentEncoderIsOpus())
    return -1;
  return codec_manager_.CurrentEncoder()->SetDtx(false) ? 0 : -1;
}

int AudioCodingModuleImpl::PlayoutTimestamp(uint32_t* timestamp) {
  return receiver_.GetPlayoutTimestamp(timestamp) ? 0 : -1;
}

bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const {
  if (!codec_manager_.CurrentEncoder()) {
    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                 "%s failed: No send codec is registered.", caller_name);
    return false;
  }
  return true;
}

int AudioCodingModuleImpl::UnregisterReceiveCodec(uint8_t payload_type) {
  return receiver_.RemoveCodec(payload_type);
}

int AudioCodingModuleImpl::SetInitialPlayoutDelay(int delay_ms) {
  {
    CriticalSectionScoped lock(acm_crit_sect_.get());
    // Initialize receiver, if it is not initialized. Otherwise, initial delay
    // is reset upon initialization of the receiver.
    if (!receiver_initialized_)
      InitializeReceiverSafe();
  }
  return receiver_.SetInitialDelay(delay_ms);
}

int AudioCodingModuleImpl::EnableNack(size_t max_nack_list_size) {
  return receiver_.EnableNack(max_nack_list_size);
}

void AudioCodingModuleImpl::DisableNack() {
  receiver_.DisableNack();
}

std::vector<uint16_t> AudioCodingModuleImpl::GetNackList(
    int64_t round_trip_time_ms) const {
  return receiver_.GetNackList(round_trip_time_ms);
}

int AudioCodingModuleImpl::LeastRequiredDelayMs() const {
  return receiver_.LeastRequiredDelayMs();
}

void AudioCodingModuleImpl::GetDecodingCallStatistics(
      AudioDecodingCallStats* call_stats) const {
  receiver_.GetDecodingCallStatistics(call_stats);
}

}  // namespace acm2
}  // namespace webrtc