modules/audio_coding/codecs/g722/audio_decoder_g722.cc

/*
 *  Copyright (c) 2015 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 "modules/audio_coding/codecs/g722/audio_decoder_g722.h"

#include <string.h>

#include <utility>

#include "modules/audio_coding/codecs/g722/g722_interface.h"
#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
#include "rtc_base/checks.h"

namespace webrtc {

AudioDecoderG722Impl::AudioDecoderG722Impl() {
  WebRtcG722_CreateDecoder(&dec_state_);
  WebRtcG722_DecoderInit(dec_state_);
}

AudioDecoderG722Impl::~AudioDecoderG722Impl() {
  WebRtcG722_FreeDecoder(dec_state_);
}

bool AudioDecoderG722Impl::HasDecodePlc() const {
  return false;
}

int AudioDecoderG722Impl::DecodeInternal(const uint8_t* encoded,
                                         size_t encoded_len,
                                         int sample_rate_hz,
                                         int16_t* decoded,
                                         SpeechType* speech_type) {
  RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
  int16_t temp_type = 1;  // Default is speech.
  size_t ret =
      WebRtcG722_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
  *speech_type = ConvertSpeechType(temp_type);
  return static_cast<int>(ret);
}

void AudioDecoderG722Impl::Reset() {
  WebRtcG722_DecoderInit(dec_state_);
}

std::vector<AudioDecoder::ParseResult> AudioDecoderG722Impl::ParsePayload(
    rtc::Buffer&& payload,
    uint32_t timestamp) {
  return LegacyEncodedAudioFrame::SplitBySamples(this, std::move(payload),
                                                 timestamp, 8, 16);
}

int AudioDecoderG722Impl::PacketDuration(const uint8_t* encoded,
                                         size_t encoded_len) const {
  // 1/2 encoded byte per sample per channel.
  return static_cast<int>(2 * encoded_len / Channels());
}

int AudioDecoderG722Impl::PacketDurationRedundant(const uint8_t* encoded,
                                                  size_t encoded_len) const {
  return PacketDuration(encoded, encoded_len);
}

int AudioDecoderG722Impl::SampleRateHz() const {
  return 16000;
}

size_t AudioDecoderG722Impl::Channels() const {
  return 1;
}

AudioDecoderG722StereoImpl::AudioDecoderG722StereoImpl() {
  WebRtcG722_CreateDecoder(&dec_state_left_);
  WebRtcG722_CreateDecoder(&dec_state_right_);
  WebRtcG722_DecoderInit(dec_state_left_);
  WebRtcG722_DecoderInit(dec_state_right_);
}

AudioDecoderG722StereoImpl::~AudioDecoderG722StereoImpl() {
  WebRtcG722_FreeDecoder(dec_state_left_);
  WebRtcG722_FreeDecoder(dec_state_right_);
}

int AudioDecoderG722StereoImpl::DecodeInternal(const uint8_t* encoded,
                                               size_t encoded_len,
                                               int sample_rate_hz,
                                               int16_t* decoded,
                                               SpeechType* speech_type) {
  RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
  // Adjust the encoded length down to ensure the same number of samples in each
  // channel.
  const size_t encoded_len_adjusted = PacketDuration(encoded, encoded_len) *
                                      Channels() /
                                      2;  // 1/2 byte per sample per channel
  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_adjusted];
  SplitStereoPacket(encoded, encoded_len_adjusted, encoded_deinterleaved);
  // Decode left and right.
  size_t decoded_len =
      WebRtcG722_Decode(dec_state_left_, encoded_deinterleaved,
                        encoded_len_adjusted / 2, decoded, &temp_type);
  size_t ret = WebRtcG722_Decode(
      dec_state_right_, &encoded_deinterleaved[encoded_len_adjusted / 2],
      encoded_len_adjusted / 2, &decoded[decoded_len], &temp_type);
  if (ret == decoded_len) {
    ret += decoded_len;  // Return total number of samples.
    // Interleave output.
    for (size_t k = ret / 2; k < ret; k++) {
      int16_t temp = decoded[k];
      memmove(&decoded[2 * k - ret + 2], &decoded[2 * k - ret + 1],
              (ret - k - 1) * sizeof(int16_t));
      decoded[2 * k - ret + 1] = temp;
    }
  }
  *speech_type = ConvertSpeechType(temp_type);
  delete[] encoded_deinterleaved;
  return static_cast<int>(ret);
}

int AudioDecoderG722StereoImpl::PacketDuration(const uint8_t* encoded,
                                               size_t encoded_len) const {
  // 1/2 encoded byte per sample per channel. Make sure the length represents
  // an equal number of bytes per channel. Otherwise, we cannot de-interleave
  // the encoded data later.
  return static_cast<int>(2 * (encoded_len / Channels()));
}

int AudioDecoderG722StereoImpl::SampleRateHz() const {
  return 16000;
}

size_t AudioDecoderG722StereoImpl::Channels() const {
  return 2;
}

void AudioDecoderG722StereoImpl::Reset() {
  WebRtcG722_DecoderInit(dec_state_left_);
  WebRtcG722_DecoderInit(dec_state_right_);
}

std::vector<AudioDecoder::ParseResult> AudioDecoderG722StereoImpl::ParsePayload(
    rtc::Buffer&& payload,
    uint32_t timestamp) {
  return LegacyEncodedAudioFrame::SplitBySamples(this, std::move(payload),
                                                 timestamp, 2 * 8, 16);
}

// Split the stereo packet and place left and right channel after each other
// in the output array.
void AudioDecoderG722StereoImpl::SplitStereoPacket(
    const uint8_t* encoded,
    size_t encoded_len,
    uint8_t* encoded_deinterleaved) {
  // 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;
  }
}

}  // namespace webrtc