text: /*
   *  Copyright (c) 2016 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/neteq/tools/fake_decode_from_file.h"
  
  #include "modules/rtp_rtcp/source/byte_io.h"
  #include "rtc_base/checks.h"
  #include "rtc_base/numerics/safe_conversions.h"
  
  namespace webrtc {
  namespace test {
  
  namespace {
  
  class FakeEncodedFrame : public AudioDecoder::EncodedAudioFrame {
   public:
    FakeEncodedFrame(AudioDecoder* decoder, rtc::Buffer&& payload)
        : decoder_(decoder), payload_(std::move(payload)) {}
  
    size_t Duration() const override {
      const int ret = decoder_->PacketDuration(payload_.data(), payload_.size());
      return ret < 0 ? 0 : static_cast<size_t>(ret);
    }
  
    absl::optional<DecodeResult> Decode(
        rtc::ArrayView<int16_t> decoded) const override {
      auto speech_type = AudioDecoder::kSpeech;
      const int ret = decoder_->Decode(
          payload_.data(), payload_.size(), decoder_->SampleRateHz(),
          decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
      return ret < 0 ? absl::nullopt
                     : absl::optional<DecodeResult>(
                           {static_cast<size_t>(ret), speech_type});
    }
  
    // This is to mimic OpusFrame.
    bool IsDtxPacket() const override {
      uint32_t original_payload_size_bytes =
          ByteReader<uint32_t>::ReadLittleEndian(&payload_.data()[8]);
      return original_payload_size_bytes <= 2;
    }
  
   private:
    AudioDecoder* const decoder_;
    const rtc::Buffer payload_;
  };
  
  }  // namespace
  
  std::vector<AudioDecoder::ParseResult> FakeDecodeFromFile::ParsePayload(
      rtc::Buffer&& payload,
      uint32_t timestamp) {
    std::vector<ParseResult> results;
    std::unique_ptr<EncodedAudioFrame> frame(
        new FakeEncodedFrame(this, std::move(payload)));
    results.emplace_back(timestamp, 0, std::move(frame));
    return results;
  }
  
  int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded,
                                         size_t encoded_len,
                                         int sample_rate_hz,
                                         int16_t* decoded,
                                         SpeechType* speech_type) {
    RTC_DCHECK_EQ(sample_rate_hz, SampleRateHz());
  
    const int samples_to_decode = PacketDuration(encoded, encoded_len);
    const int total_samples_to_decode = samples_to_decode * (stereo_ ? 2 : 1);
  
    if (encoded_len == 0) {
      // Decoder is asked to produce codec-internal comfort noise.
      RTC_DCHECK(!encoded);  // NetEq always sends nullptr in this case.
      RTC_DCHECK(cng_mode_);
      RTC_DCHECK_GT(total_samples_to_decode, 0);
      std::fill_n(decoded, total_samples_to_decode, 0);
      *speech_type = kComfortNoise;
      return rtc::dchecked_cast<int>(total_samples_to_decode);
    }
  
    RTC_CHECK_GE(encoded_len, 12);
    uint32_t timestamp_to_decode =
        ByteReader<uint32_t>::ReadLittleEndian(encoded);
  
    if (next_timestamp_from_input_ &&
        timestamp_to_decode != *next_timestamp_from_input_) {
      // A gap in the timestamp sequence is detected. Skip the same number of
      // samples from the file.
      uint32_t jump = timestamp_to_decode - *next_timestamp_from_input_;
      RTC_CHECK(input_->Seek(jump));
    }
  
    next_timestamp_from_input_ = timestamp_to_decode + samples_to_decode;
  
    uint32_t original_payload_size_bytes =
        ByteReader<uint32_t>::ReadLittleEndian(&encoded[8]);
    if (original_payload_size_bytes <= 2) {
      // This is a comfort noise payload.
      RTC_DCHECK_GT(total_samples_to_decode, 0);
      std::fill_n(decoded, total_samples_to_decode, 0);
      *speech_type = kComfortNoise;
      cng_mode_ = true;
      return rtc::dchecked_cast<int>(total_samples_to_decode);
    }
  
    cng_mode_ = false;
    RTC_CHECK(input_->Read(static_cast<size_t>(samples_to_decode), decoded));
  
    if (stereo_) {
      InputAudioFile::DuplicateInterleaved(decoded, samples_to_decode, 2,
                                           decoded);
    }
  
    *speech_type = kSpeech;
    last_decoded_length_ = samples_to_decode;
    return rtc::dchecked_cast<int>(total_samples_to_decode);
  }
  
  int FakeDecodeFromFile::PacketDuration(const uint8_t* encoded,
                                         size_t encoded_len) const {
    const uint32_t original_payload_size_bytes =
        encoded_len < 8 + sizeof(uint32_t)
            ? 0
            : ByteReader<uint32_t>::ReadLittleEndian(&encoded[8]);
    const uint32_t samples_to_decode =
        encoded_len < 4 + sizeof(uint32_t)
            ? 0
            : ByteReader<uint32_t>::ReadLittleEndian(&encoded[4]);
    if (  // Decoder is asked to produce codec-internal comfort noise
        encoded_len == 0 ||
        // Comfort noise payload
        original_payload_size_bytes <= 2 || samples_to_decode == 0 ||
        // Erroneous duration since it is not a multiple of 10ms
        samples_to_decode % rtc::CheckedDivExact(SampleRateHz(), 100) != 0) {
      if (last_decoded_length_ > 0) {
        // Use length of last decoded packet.
        return rtc::dchecked_cast<int>(last_decoded_length_);
      } else {
        // This is the first packet to decode, and we do not know the length of
        // it. Set it to 10 ms.
        return rtc::CheckedDivExact(SampleRateHz(), 100);
      }
    }
    return samples_to_decode;
  }
  
  void FakeDecodeFromFile::PrepareEncoded(uint32_t timestamp,
                                          size_t samples,
                                          size_t original_payload_size_bytes,
                                          rtc::ArrayView<uint8_t> encoded) {
    RTC_CHECK_GE(encoded.size(), 12);
    ByteWriter<uint32_t>::WriteLittleEndian(&encoded[0], timestamp);
    ByteWriter<uint32_t>::WriteLittleEndian(&encoded[4],
                                            rtc::checked_cast<uint32_t>(samples));
    ByteWriter<uint32_t>::WriteLittleEndian(
        &encoded[8], rtc::checked_cast<uint32_t>(original_payload_size_bytes));
  }
  
  }  // namespace test
  }  // namespace webrtc
  