| Index: webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
|
| diff --git a/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc b/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
|
| index 9ca5fb810bd64f21298680634793cc1b53a55ae5..29beed5644f31402bda6d74d75fde25b676d4ab3 100644
|
| --- a/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
|
| +++ b/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
|
| @@ -19,14 +19,37 @@ namespace test {
|
|
|
| int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded,
|
| size_t encoded_len,
|
| - int /*sample_rate_hz*/,
|
| + int sample_rate_hz,
|
| int16_t* decoded,
|
| SpeechType* speech_type) {
|
| - RTC_CHECK_GE(encoded_len, 8u);
|
| + 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(last_decoded_length_, 0u);
|
| + std::fill_n(decoded, last_decoded_length_, 0);
|
| + *speech_type = kComfortNoise;
|
| + return last_decoded_length_;
|
| + }
|
| +
|
| + RTC_CHECK_GE(encoded_len, 12u);
|
| uint32_t timestamp_to_decode =
|
| ByteReader<uint32_t>::ReadLittleEndian(encoded);
|
| uint32_t samples_to_decode =
|
| ByteReader<uint32_t>::ReadLittleEndian(&encoded[4]);
|
| + if (samples_to_decode == 0) {
|
| + // Number of samples in packet is unknown.
|
| + if (last_decoded_length_ > 0) {
|
| + // Use length of last decoded packet, but since this is the total for all
|
| + // channels, we have to divide by 2 in the stereo case.
|
| + samples_to_decode = rtc::CheckedDivExact(
|
| + last_decoded_length_, static_cast<size_t>(stereo_ ? 2uL : 1uL));
|
| + } else {
|
| + // This is the first packet to decode, and we do not know the length of
|
| + // it. Set it to 10 ms.
|
| + samples_to_decode = rtc::CheckedDivExact(sample_rate_hz, 100);
|
| + }
|
| + }
|
|
|
| if (next_timestamp_from_input_ &&
|
| timestamp_to_decode != *next_timestamp_from_input_) {
|
| @@ -36,10 +59,23 @@ int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded,
|
| RTC_CHECK(input_->Seek(jump));
|
| }
|
|
|
| - RTC_CHECK(input_->Read(static_cast<size_t>(samples_to_decode), decoded));
|
| next_timestamp_from_input_ =
|
| rtc::Optional<uint32_t>(timestamp_to_decode + samples_to_decode);
|
|
|
| + uint32_t original_payload_size_bytes =
|
| + ByteReader<uint32_t>::ReadLittleEndian(&encoded[8]);
|
| + if (original_payload_size_bytes == 1) {
|
| + // This is a comfort noise payload.
|
| + RTC_DCHECK_GT(last_decoded_length_, 0u);
|
| + std::fill_n(decoded, last_decoded_length_, 0);
|
| + *speech_type = kComfortNoise;
|
| + cng_mode_ = true;
|
| + return last_decoded_length_;
|
| + }
|
| +
|
| + 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);
|
| @@ -47,16 +83,19 @@ int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded,
|
| }
|
|
|
| *speech_type = kSpeech;
|
| - return samples_to_decode;
|
| + return last_decoded_length_ = 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(), 8u);
|
| + RTC_CHECK_GE(encoded.size(), 12u);
|
| 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
|
|
|