text: /*
   *  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/voice_engine/coder.h"
  
  #include "webrtc/common_types.h"
  #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
  #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
  #include "webrtc/modules/include/module_common_types.h"
  
  namespace webrtc {
  namespace {
  AudioCodingModule::Config GetAcmConfig(uint32_t id) {
    AudioCodingModule::Config config;
    // This class does not handle muted output.
    config.neteq_config.enable_muted_state = false;
    config.id = id;
    config.decoder_factory = CreateBuiltinAudioDecoderFactory();
    return config;
  }
  }  // namespace
  
  AudioCoder::AudioCoder(uint32_t instance_id)
      : acm_(AudioCodingModule::Create(GetAcmConfig(instance_id))),
        receive_codec_(),
        encode_timestamp_(0),
        encoded_data_(nullptr),
        encoded_length_in_bytes_(0),
        decode_timestamp_(0) {
    acm_->InitializeReceiver();
    acm_->RegisterTransportCallback(this);
  }
  
  AudioCoder::~AudioCoder() {}
  
  int32_t AudioCoder::SetEncodeCodec(const CodecInst& codec_inst) {
    const bool success = codec_manager_.RegisterEncoder(codec_inst) &&
                         codec_manager_.MakeEncoder(&rent_a_codec_, acm_.get());
    return success ? 0 : -1;
  }
  
  int32_t AudioCoder::SetDecodeCodec(const CodecInst& codec_inst) {
    if (!acm_->RegisterReceiveCodec(codec_inst.pltype,
                                    CodecInstToSdp(codec_inst))) {
      return -1;
    }
    memcpy(&receive_codec_, &codec_inst, sizeof(CodecInst));
    return 0;
  }
  
  int32_t AudioCoder::Decode(AudioFrame* decoded_audio,
                             uint32_t samp_freq_hz,
                             const int8_t* incoming_payload,
                             size_t payload_length) {
    if (payload_length > 0) {
      const uint8_t payload_type = receive_codec_.pltype;
      decode_timestamp_ += receive_codec_.pacsize;
      if (acm_->IncomingPayload((const uint8_t*)incoming_payload, payload_length,
                                payload_type, decode_timestamp_) == -1) {
        return -1;
      }
    }
    bool muted;
    int32_t ret =
        acm_->PlayoutData10Ms((uint16_t)samp_freq_hz, decoded_audio, &muted);
    RTC_DCHECK(!muted);
    return ret;
  }
  
  int32_t AudioCoder::PlayoutData(AudioFrame* decoded_audio,
                                  uint16_t samp_freq_hz) {
    bool muted;
    int32_t ret = acm_->PlayoutData10Ms(samp_freq_hz, decoded_audio, &muted);
    RTC_DCHECK(!muted);
    return ret;
  }
  
  int32_t AudioCoder::Encode(const AudioFrame& audio,
                             int8_t* encoded_data,
                             size_t* encoded_length_in_bytes) {
    // Fake a timestamp in case audio doesn't contain a correct timestamp.
    // Make a local copy of the audio frame since audio is const
    AudioFrame audio_frame;
    audio_frame.CopyFrom(audio);
    audio_frame.timestamp_ = encode_timestamp_;
    encode_timestamp_ += static_cast<uint32_t>(audio_frame.samples_per_channel_);
  
    // For any codec with a frame size that is longer than 10 ms the encoded
    // length in bytes should be zero until a a full frame has been encoded.
    encoded_length_in_bytes_ = 0;
    if (acm_->Add10MsData((AudioFrame&)audio_frame) == -1) {
      return -1;
    }
    encoded_data_ = encoded_data;
    *encoded_length_in_bytes = encoded_length_in_bytes_;
    return 0;
  }
  
  int32_t AudioCoder::SendData(FrameType /* frame_type */,
                               uint8_t /* payload_type */,
                               uint32_t /* time_stamp */,
                               const uint8_t* payload_data,
                               size_t payload_size,
                               const RTPFragmentationHeader* /* fragmentation*/) {
    memcpy(encoded_data_, payload_data, sizeof(uint8_t) * payload_size);
    encoded_length_in_bytes_ = payload_size;
    return 0;
  }
  
  }  // namespace webrtc
  