Removing AudioCoding class, a.k.a the new ACM API

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

 /*
  *  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.
  */
 
@@ skipping 785 matching lines @@
 int AudioCodingModuleImpl::LeastRequiredDelayMs() const {
   return receiver_.LeastRequiredDelayMs();
 }
 
 void AudioCodingModuleImpl::GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const {
   receiver_.GetDecodingCallStatistics(call_stats);
 }
 
 }  // namespace acm2
-
-AudioCodingImpl::AudioCodingImpl(const Config& config) {
-  AudioCodingModule::Config config_old = config.ToOldConfig();
-  acm_old_.reset(new acm2::AudioCodingModuleImpl(config_old));
-  acm_old_->RegisterTransportCallback(config.transport);
-  acm_old_->RegisterVADCallback(config.vad_callback);
-  if (config.initial_playout_delay_ms > 0) {
-    acm_old_->SetInitialPlayoutDelay(config.initial_playout_delay_ms);
-  }
-  playout_frequency_hz_ = config.playout_frequency_hz;
-}
-
-AudioCodingImpl::~AudioCodingImpl() = default;
-
-bool AudioCodingImpl::RegisterSendCodec(AudioEncoder* send_codec) {
-  FATAL() << "Not implemented yet.";
-  return false;
-}
-
-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.";
-  return reinterpret_cast<const AudioEncoder*>(NULL);
-}
-
-const CodecInst* AudioCodingImpl::GetSenderCodecInst() {
-  if (acm_old_->SendCodec(&current_send_codec_) != 0) {
-    return NULL;
-  }
-  return &current_send_codec_;
-}
-
-int AudioCodingImpl::Add10MsAudio(const AudioFrame& audio_frame) {
-  acm2::AudioCodingModuleImpl::InputData input_data;
-  CriticalSectionScoped lock(acm_old_->acm_crit_sect_.get());
-  if (acm_old_->Add10MsDataInternal(audio_frame, &input_data) != 0)
-    return -1;
-  return acm_old_->Encode(input_data);
-}
-
-const ReceiverInfo* AudioCodingImpl::GetReceiverInfo() const {
-  FATAL() << "Not implemented yet.";
-  return reinterpret_cast<const ReceiverInfo*>(NULL);
-}
-
-bool AudioCodingImpl::RegisterReceiveCodec(AudioDecoder* receive_codec) {
-  FATAL() << "Not implemented yet.";
-  return false;
-}
-
-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,
-                                   size_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,
-                                    size_t payload_len_byte,
-                                    uint8_t payload_type,
-                                    uint32_t timestamp) {
-  FATAL() << "Not implemented yet.";
-  return false;
-}
-
-bool AudioCodingImpl::SetMinimumPlayoutDelay(int time_ms) {
-  FATAL() << "Not implemented yet.";
-  return false;
-}
-
-bool AudioCodingImpl::SetMaximumPlayoutDelay(int time_ms) {
-  FATAL() << "Not implemented yet.";
-  return false;
-}
-
-int AudioCodingImpl::LeastRequiredDelayMs() const {
-  FATAL() << "Not implemented yet.";
-  return -1;
-}
-
-bool AudioCodingImpl::PlayoutTimestamp(uint32_t* timestamp) {
-  FATAL() << "Not implemented yet.";
-  return false;
-}
-
-bool AudioCodingImpl::Get10MsAudio(AudioFrame* audio_frame) {
-  return acm_old_->PlayoutData10Ms(playout_frequency_hz_, audio_frame) == 0;
-}
-
-bool AudioCodingImpl::GetNetworkStatistics(
-    NetworkStatistics* network_statistics) {
-  FATAL() << "Not implemented yet.";
-  return false;
-}
-
-bool AudioCodingImpl::EnableNack(size_t max_nack_list_size) {
-  FATAL() << "Not implemented yet.";
-  return false;
-}
-
-void AudioCodingImpl::DisableNack() {
-  // A bug in the linker of Visual Studio 2013 Update 3 prevent us from using
-  // FATAL() here, if we do so then the linker hang when the WPO is turned on.
-  // TODO(sebmarchand): Re-evaluate this when we upgrade the toolchain.
-}
-
-bool AudioCodingImpl::SetVad(bool enable_dtx,
-                             bool enable_vad,
-                             ACMVADMode vad_mode) {
-  return acm_old_->SetVAD(enable_dtx, enable_vad, vad_mode) == 0;
-}
-
-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) {
-    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;
-#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;
-#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;
-    case acm2::ACMCodecDB::kAVT:
-      *codec_name = "telephone-event";
-      *sample_rate_hz = 8000;
-      *channels = 1;
-      break;
-    default:
-      FATAL() << "Codec type " << codec_type << " not supported.";
-  }
-  return true;
-}
-
 }  // namespace webrtc