webrtc/modules/audio_coding/neteq/audio_decoder_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.
 */

#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"

#include <assert.h>
#include <string.h>  // memmove

#include "webrtc/base/checks.h"
#ifdef WEBRTC_CODEC_CELT
#include "webrtc/modules/audio_coding/codecs/celt/include/celt_interface.h"
#endif
#include "webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h"
#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
#ifdef WEBRTC_CODEC_G722
#include "webrtc/modules/audio_coding/codecs/g722/include/g722_interface.h"
#endif
#ifdef WEBRTC_CODEC_ILBC
#include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
#endif
#ifdef WEBRTC_CODEC_ISACFX
#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
#endif
#ifdef WEBRTC_CODEC_ISAC
#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
#endif
#ifdef WEBRTC_CODEC_OPUS
#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
#endif
#ifdef WEBRTC_CODEC_PCM16
#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
#endif

namespace webrtc {

// PCMu
int AudioDecoderPcmU::Decode(const uint8_t* encoded, size_t encoded_len,
                              int16_t* decoded, SpeechType* speech_type) {
  int16_t temp_type = 1;  // Default is speech.
  int16_t ret = WebRtcG711_DecodeU(
      reinterpret_cast<int16_t*>(const_cast<uint8_t*>(encoded)),
      static_cast<int16_t>(encoded_len), decoded, &temp_type);
  *speech_type = ConvertSpeechType(temp_type);
  return ret;
}

int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded,
                                     size_t encoded_len) {
  // One encoded byte per sample per channel.
  return static_cast<int>(encoded_len / channels_);
}

// PCMa
int AudioDecoderPcmA::Decode(const uint8_t* encoded, size_t encoded_len,
                              int16_t* decoded, SpeechType* speech_type) {
  int16_t temp_type = 1;  // Default is speech.
  int16_t ret = WebRtcG711_DecodeA(
      reinterpret_cast<int16_t*>(const_cast<uint8_t*>(encoded)),
      static_cast<int16_t>(encoded_len), decoded, &temp_type);
  *speech_type = ConvertSpeechType(temp_type);
  return ret;
}

int AudioDecoderPcmA::PacketDuration(const uint8_t* encoded,
                                     size_t encoded_len) {
  // One encoded byte per sample per channel.
  return static_cast<int>(encoded_len / channels_);
}

// PCM16B
#ifdef WEBRTC_CODEC_PCM16
AudioDecoderPcm16B::AudioDecoderPcm16B() {}

int AudioDecoderPcm16B::Decode(const uint8_t* encoded, size_t encoded_len,
                               int16_t* decoded, SpeechType* speech_type) {
  int16_t temp_type = 1;  // Default is speech.
  int16_t ret = WebRtcPcm16b_DecodeW16(
      reinterpret_cast<int16_t*>(const_cast<uint8_t*>(encoded)),
      static_cast<int16_t>(encoded_len), decoded, &temp_type);
  *speech_type = ConvertSpeechType(temp_type);
  return ret;
}

int AudioDecoderPcm16B::PacketDuration(const uint8_t* encoded,
                                       size_t encoded_len) {
  // Two encoded byte per sample per channel.
  return static_cast<int>(encoded_len / (2 * channels_));
}

AudioDecoderPcm16BMultiCh::AudioDecoderPcm16BMultiCh(int num_channels) {
  DCHECK(num_channels > 0);
  channels_ = num_channels;
}
#endif

// iLBC
#ifdef WEBRTC_CODEC_ILBC
AudioDecoderIlbc::AudioDecoderIlbc() {
  WebRtcIlbcfix_DecoderCreate(reinterpret_cast<iLBC_decinst_t**>(&state_));
}

AudioDecoderIlbc::~AudioDecoderIlbc() {
  WebRtcIlbcfix_DecoderFree(static_cast<iLBC_decinst_t*>(state_));
}

int AudioDecoderIlbc::Decode(const uint8_t* encoded, size_t encoded_len,
                             int16_t* decoded, SpeechType* speech_type) {
  int16_t temp_type = 1;  // Default is speech.
  int16_t ret = WebRtcIlbcfix_Decode(static_cast<iLBC_decinst_t*>(state_),
                                     reinterpret_cast<const int16_t*>(encoded),
                                     static_cast<int16_t>(encoded_len), decoded,
                                     &temp_type);
  *speech_type = ConvertSpeechType(temp_type);
  return ret;
}

int AudioDecoderIlbc::DecodePlc(int num_frames, int16_t* decoded) {
  return WebRtcIlbcfix_NetEqPlc(static_cast<iLBC_decinst_t*>(state_),
                                decoded, num_frames);
}

int AudioDecoderIlbc::Init() {
  return WebRtcIlbcfix_Decoderinit30Ms(static_cast<iLBC_decinst_t*>(state_));
}
#endif

// iSAC float
#ifdef WEBRTC_CODEC_ISAC
AudioDecoderIsac::AudioDecoderIsac(int decode_sample_rate_hz) {
  DCHECK(decode_sample_rate_hz == 16000 || decode_sample_rate_hz == 32000);
  WebRtcIsac_Create(reinterpret_cast<ISACStruct**>(&state_));
  WebRtcIsac_SetDecSampRate(static_cast<ISACStruct*>(state_),
                            decode_sample_rate_hz);
}

AudioDecoderIsac::~AudioDecoderIsac() {
  WebRtcIsac_Free(static_cast<ISACStruct*>(state_));
}

int AudioDecoderIsac::Decode(const uint8_t* encoded, size_t encoded_len,
                             int16_t* decoded, SpeechType* speech_type) {
  int16_t temp_type = 1;  // Default is speech.
  int16_t ret = WebRtcIsac_Decode(static_cast<ISACStruct*>(state_),
                                  encoded,
                                  static_cast<int16_t>(encoded_len), decoded,
                                  &temp_type);
  *speech_type = ConvertSpeechType(temp_type);
  return ret;
}

int AudioDecoderIsac::DecodeRedundant(const uint8_t* encoded,
                                      size_t encoded_len, int16_t* decoded,
                                      SpeechType* speech_type) {
  int16_t temp_type = 1;  // Default is speech.
  int16_t ret = WebRtcIsac_DecodeRcu(static_cast<ISACStruct*>(state_),
                                     encoded,
                                     static_cast<int16_t>(encoded_len), decoded,
                                     &temp_type);
  *speech_type = ConvertSpeechType(temp_type);
  return ret;
}

int AudioDecoderIsac::DecodePlc(int num_frames, int16_t* decoded) {
  return WebRtcIsac_DecodePlc(static_cast<ISACStruct*>(state_),
                                 decoded, num_frames);
}

int AudioDecoderIsac::Init() {
  return WebRtcIsac_DecoderInit(static_cast<ISACStruct*>(state_));
}

int AudioDecoderIsac::IncomingPacket(const uint8_t* payload,
                                     size_t payload_len,
                                     uint16_t rtp_sequence_number,
                                     uint32_t rtp_timestamp,
                                     uint32_t arrival_timestamp) {
  return WebRtcIsac_UpdateBwEstimate(static_cast<ISACStruct*>(state_),
                                     payload,
                                     static_cast<int32_t>(payload_len),
                                     rtp_sequence_number,
                                     rtp_timestamp,
                                     arrival_timestamp);
}

int AudioDecoderIsac::ErrorCode() {
  return WebRtcIsac_GetErrorCode(static_cast<ISACStruct*>(state_));
}
#endif

// iSAC fix
#ifdef WEBRTC_CODEC_ISACFX
AudioDecoderIsacFix::AudioDecoderIsacFix() {
  WebRtcIsacfix_Create(reinterpret_cast<ISACFIX_MainStruct**>(&state_));
}

AudioDecoderIsacFix::~AudioDecoderIsacFix() {
  WebRtcIsacfix_Free(static_cast<ISACFIX_MainStruct*>(state_));
}

int AudioDecoderIsacFix::Decode(const uint8_t* encoded, size_t encoded_len,
                                int16_t* decoded, SpeechType* speech_type) {
  int16_t temp_type = 1;  // Default is speech.
  int16_t ret = WebRtcIsacfix_Decode(static_cast<ISACFIX_MainStruct*>(state_),
                                     encoded,
                                     static_cast<int16_t>(encoded_len), decoded,
                                     &temp_type);
  *speech_type = ConvertSpeechType(temp_type);
  return ret;
}

int AudioDecoderIsacFix::Init() {
  return WebRtcIsacfix_DecoderInit(static_cast<ISACFIX_MainStruct*>(state_));
}

int AudioDecoderIsacFix::IncomingPacket(const uint8_t* payload,
                                        size_t payload_len,
                                        uint16_t rtp_sequence_number,
                                        uint32_t rtp_timestamp,
                                        uint32_t arrival_timestamp) {
  return WebRtcIsacfix_UpdateBwEstimate(
      static_cast<ISACFIX_MainStruct*>(state_),
      payload,
      static_cast<int32_t>(payload_len),
      rtp_sequence_number, rtp_timestamp, arrival_timestamp);
}

int AudioDecoderIsacFix::ErrorCode() {
  return WebRtcIsacfix_GetErrorCode(static_cast<ISACFIX_MainStruct*>(state_));
}
#endif

// G.722
#ifdef WEBRTC_CODEC_G722
AudioDecoderG722::AudioDecoderG722() {
  WebRtcG722_CreateDecoder(reinterpret_cast<G722DecInst**>(&state_));
}

AudioDecoderG722::~AudioDecoderG722() {
  WebRtcG722_FreeDecoder(static_cast<G722DecInst*>(state_));
}

int AudioDecoderG722::Decode(const uint8_t* encoded, size_t encoded_len,
                             int16_t* decoded, SpeechType* speech_type) {
  int16_t temp_type = 1;  // Default is speech.
  int16_t ret = WebRtcG722_Decode(
      static_cast<G722DecInst*>(state_),
      const_cast<int16_t*>(reinterpret_cast<const int16_t*>(encoded)),
      static_cast<int16_t>(encoded_len), decoded, &temp_type);
  *speech_type = ConvertSpeechType(temp_type);
  return ret;
}

int AudioDecoderG722::Init() {
  return WebRtcG722_DecoderInit(static_cast<G722DecInst*>(state_));
}

int AudioDecoderG722::PacketDuration(const uint8_t* encoded,
                                     size_t encoded_len) {
  // 1/2 encoded byte per sample per channel.
  return static_cast<int>(2 * encoded_len / channels_);
}

AudioDecoderG722Stereo::AudioDecoderG722Stereo()
    : AudioDecoderG722(),
      state_left_(state_),  // Base member |state_| is used for left channel.
      state_right_(NULL) {
  channels_ = 2;
  // |state_left_| already created by the base class AudioDecoderG722.
  WebRtcG722_CreateDecoder(reinterpret_cast<G722DecInst**>(&state_right_));
}

AudioDecoderG722Stereo::~AudioDecoderG722Stereo() {
  // |state_left_| will be freed by the base class AudioDecoderG722.
  WebRtcG722_FreeDecoder(static_cast<G722DecInst*>(state_right_));
}

int AudioDecoderG722Stereo::Decode(const uint8_t* encoded, size_t encoded_len,
                                   int16_t* decoded, SpeechType* speech_type) {
  int16_t temp_type = 1;  // Default is speech.
  // De-interleave the bit-stream into two separate payloads.
  uint8_t* encoded_deinterleaved = new uint8_t[encoded_len];
  SplitStereoPacket(encoded, encoded_len, encoded_deinterleaved);
  // Decode left and right.
  int16_t ret = WebRtcG722_Decode(
      static_cast<G722DecInst*>(state_left_),
      reinterpret_cast<int16_t*>(encoded_deinterleaved),
      static_cast<int16_t>(encoded_len / 2), decoded, &temp_type);
  if (ret >= 0) {
    int decoded_len = ret;
    ret = WebRtcG722_Decode(
      static_cast<G722DecInst*>(state_right_),
      reinterpret_cast<int16_t*>(&encoded_deinterleaved[encoded_len / 2]),
      static_cast<int16_t>(encoded_len / 2), &decoded[decoded_len], &temp_type);
    if (ret == decoded_len) {
      decoded_len += ret;
      // Interleave output.
      for (int k = decoded_len / 2; k < decoded_len; k++) {
          int16_t temp = decoded[k];
          memmove(&decoded[2 * k - decoded_len + 2],
                  &decoded[2 * k - decoded_len + 1],
                  (decoded_len - k - 1) * sizeof(int16_t));
          decoded[2 * k - decoded_len + 1] = temp;
      }
      ret = decoded_len;  // Return total number of samples.
    }
  }
  *speech_type = ConvertSpeechType(temp_type);
  delete [] encoded_deinterleaved;
  return ret;
}

int AudioDecoderG722Stereo::Init() {
  int ret = WebRtcG722_DecoderInit(static_cast<G722DecInst*>(state_right_));
  if (ret != 0) {
    return ret;
  }
  return AudioDecoderG722::Init();
}

// Split the stereo packet and place left and right channel after each other
// in the output array.
void AudioDecoderG722Stereo::SplitStereoPacket(const uint8_t* encoded,
                                               size_t encoded_len,
                                               uint8_t* encoded_deinterleaved) {
  assert(encoded);
  // Regroup the 4 bits/sample so |l1 l2| |r1 r2| |l3 l4| |r3 r4| ...,
  // where "lx" is 4 bits representing left sample number x, and "rx" right
  // sample. Two samples fit in one byte, represented with |...|.
  for (size_t i = 0; i + 1 < encoded_len; i += 2) {
    uint8_t right_byte = ((encoded[i] & 0x0F) << 4) + (encoded[i + 1] & 0x0F);
    encoded_deinterleaved[i] = (encoded[i] & 0xF0) + (encoded[i + 1] >> 4);
    encoded_deinterleaved[i + 1] = right_byte;
  }

  // Move one byte representing right channel each loop, and place it at the
  // end of the bytestream vector. After looping the data is reordered to:
  // |l1 l2| |l3 l4| ... |l(N-1) lN| |r1 r2| |r3 r4| ... |r(N-1) r(N)|,
  // where N is the total number of samples.
  for (size_t i = 0; i < encoded_len / 2; i++) {
    uint8_t right_byte = encoded_deinterleaved[i + 1];
    memmove(&encoded_deinterleaved[i + 1], &encoded_deinterleaved[i + 2],
            encoded_len - i - 2);
    encoded_deinterleaved[encoded_len - 1] = right_byte;
  }
}
#endif

// CELT
#ifdef WEBRTC_CODEC_CELT
AudioDecoderCelt::AudioDecoderCelt(int num_channels) {
  DCHECK(num_channels == 1 || num_channels == 2);
  channels_ = num_channels;
  WebRtcCelt_CreateDec(reinterpret_cast<CELT_decinst_t**>(&state_),
                       static_cast<int>(channels_));
}

AudioDecoderCelt::~AudioDecoderCelt() {
  WebRtcCelt_FreeDec(static_cast<CELT_decinst_t*>(state_));
}

int AudioDecoderCelt::Decode(const uint8_t* encoded, size_t encoded_len,
                             int16_t* decoded, SpeechType* speech_type) {
  int16_t temp_type = 1;  // Default to speech.
  int ret = WebRtcCelt_DecodeUniversal(static_cast<CELT_decinst_t*>(state_),
                                       encoded, static_cast<int>(encoded_len),
                                       decoded, &temp_type);
  *speech_type = ConvertSpeechType(temp_type);
  if (ret < 0) {
    return -1;
  }
  // Return the total number of samples.
  return ret * static_cast<int>(channels_);
}

int AudioDecoderCelt::Init() {
  return WebRtcCelt_DecoderInit(static_cast<CELT_decinst_t*>(state_));
}

bool AudioDecoderCelt::HasDecodePlc() const { return true; }

int AudioDecoderCelt::DecodePlc(int num_frames, int16_t* decoded) {
  int ret = WebRtcCelt_DecodePlc(static_cast<CELT_decinst_t*>(state_),
                                 decoded, num_frames);
  if (ret < 0) {
    return -1;
  }
  // Return the total number of samples.
  return ret * static_cast<int>(channels_);
}
#endif

// Opus
#ifdef WEBRTC_CODEC_OPUS
AudioDecoderOpus::AudioDecoderOpus(int num_channels) {
  DCHECK(num_channels == 1 || num_channels == 2);
  channels_ = num_channels;
  WebRtcOpus_DecoderCreate(reinterpret_cast<OpusDecInst**>(&state_),
                           static_cast<int>(channels_));
}

AudioDecoderOpus::~AudioDecoderOpus() {
  WebRtcOpus_DecoderFree(static_cast<OpusDecInst*>(state_));
}

int AudioDecoderOpus::Decode(const uint8_t* encoded, size_t encoded_len,
                             int16_t* decoded, SpeechType* speech_type) {
  int16_t temp_type = 1;  // Default is speech.
  int16_t ret = WebRtcOpus_DecodeNew(static_cast<OpusDecInst*>(state_), encoded,
                                     static_cast<int16_t>(encoded_len), decoded,
                                     &temp_type);
  if (ret > 0)
    ret *= static_cast<int16_t>(channels_);  // Return total number of samples.
  *speech_type = ConvertSpeechType(temp_type);
  return ret;
}

int AudioDecoderOpus::DecodeRedundant(const uint8_t* encoded,
                                      size_t encoded_len, int16_t* decoded,
                                      SpeechType* speech_type) {
  int16_t temp_type = 1;  // Default is speech.
  int16_t ret = WebRtcOpus_DecodeFec(static_cast<OpusDecInst*>(state_), encoded,
                                     static_cast<int16_t>(encoded_len), decoded,
                                     &temp_type);
  if (ret > 0)
    ret *= static_cast<int16_t>(channels_);  // Return total number of samples.
  *speech_type = ConvertSpeechType(temp_type);
  return ret;
}

int AudioDecoderOpus::Init() {
  return WebRtcOpus_DecoderInitNew(static_cast<OpusDecInst*>(state_));
}

int AudioDecoderOpus::PacketDuration(const uint8_t* encoded,
                                     size_t encoded_len) {
  return WebRtcOpus_DurationEst(static_cast<OpusDecInst*>(state_),
                                encoded, static_cast<int>(encoded_len));
}

int AudioDecoderOpus::PacketDurationRedundant(const uint8_t* encoded,
                                              size_t encoded_len) const {
  return WebRtcOpus_FecDurationEst(encoded, static_cast<int>(encoded_len));
}

bool AudioDecoderOpus::PacketHasFec(const uint8_t* encoded,
                                    size_t encoded_len) const {
  int fec;
  fec = WebRtcOpus_PacketHasFec(encoded, static_cast<int>(encoded_len));
  return (fec == 1);
}
#endif

AudioDecoderCng::AudioDecoderCng() {
  WebRtcCng_CreateDec(reinterpret_cast<CNG_dec_inst**>(&state_));
  assert(state_);
}

AudioDecoderCng::~AudioDecoderCng() {
  if (state_) {
    WebRtcCng_FreeDec(static_cast<CNG_dec_inst*>(state_));
  }
}

int AudioDecoderCng::Init() {
  assert(state_);
  return WebRtcCng_InitDec(static_cast<CNG_dec_inst*>(state_));
}

}  // namespace webrtc