audio_coding: remove "main" directory

webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc

Index: webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
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
deleted file mode 100644
index 5d18bda00c3a5e1b9aa12fd5791479e560bbaea4..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ /dev/null
@@ -1,786 +0,0 @@
-/*
- *  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/include/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, rtc::ArrayView<const int16_t>(
-                         input_data.audio, input_data.audio_channel *
-                                               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.
-rtc::Optional<CodecInst> AudioCodingModuleImpl::SendCodec() const {
-  CriticalSectionScoped lock(acm_crit_sect_.get());
-  return codec_manager_.GetCodecInst();
-}
-
-// 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 {
-  const auto last_packet_sample_rate = receiver_.last_packet_sample_rate_hz();
-  return last_packet_sample_rate ? *last_packet_sample_rate
-                                 : receiver_.last_output_sample_rate_hz();
-}
-
-// Get current playout frequency.
-int AudioCodingModuleImpl::PlayoutFrequency() const {
-  WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
-               "PlayoutFrequency()");
-  return receiver_.last_output_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 (!RentACodec::IsPayloadTypeValid(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 (!RentACodec::IsPayloadTypeValid(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,
-      rtc::ArrayView<const uint8_t>(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::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